UAD PLUG-INS MANUAL SOFTWARE VERSION 7.5 Manual version 140211 Universal Audio, Inc. 4585 Scotts Valley Drive Scotts Valley, CA 95066-4517 www.uaudio.
NOTICES Disclaimer This manual provides general information, preparation for use, installation and operating instructions for the Universal Audio UAD Powered Plug-Ins. The information contained in this manual is subject to change without notice. Universal Audio, Inc. makes no warranties of any kind with regard to this manual, or the product(s) it refers to, including, but not limited to, the implied warranties of merchantability and fitness for a particular purpose. Universal Audio, Inc.
TABLE OF CONTENTS Chapter 1. Documentation & Support . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 14 UAD Documentation Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 14 Customer Support . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 17 Chapter 2. Ampex ATR-102 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
TABLE OF CONTENTS Design Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 81 Cooper Time Cube Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 81 Channel Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 84 Cooper Time Cube Hardware . . . . . . . . . . . . . . . . . . . . . . .
TABLE OF CONTENTS Channel Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 121 Global Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 124 FATSO Sr. Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 125 Chapter 11. EMT 140 Plate Reverb . . . . . . . . . . . . . . . . .
TABLE OF CONTENTS Harrison 32C Latency . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 177 Chapter 16. Helios Type 69 Equalizer . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 178 Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 178 Helios Type 69 Screenshot . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
TABLE OF CONTENTS Massive Passive Latency . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 224 Notes from Manley Laboratories . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 224 Additional Information . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 226 Chapter 22. Moog Multimode Filter . . . . . . . . . . . . . . . . . . . . . . . . . .
TABLE OF CONTENTS Operation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 263 Neve 33609 and 33609SE Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 264 Limiter . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 264 Compressor . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
TABLE OF CONTENTS Precision De-Esser Screenshot . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 326 Precision De-Esser Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 327 Operating Tips . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 329 Chapter 32. Precision Enhancer Hz . . . . . . . . . . . . . . . . . . . . . . .
TABLE OF CONTENTS Precision Limiter Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 357 Precision Limiter Meters Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 358 Precision Limiter Latency . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 363 Chapter 37. Precision Maximizer . . . . . . . . . . . . . . . . . . . . . . . . . .
TABLE OF CONTENTS Morphing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 408 RealVerb Pro Preset Management . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 410 RealVerb Pro Preset List . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 410 Chapter 41. Boss CE-1 Chorus Ensemble . . . . . . . . . . . . . . . . . . . .
TABLE OF CONTENTS SSL G Bus Compressor Screenshot. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 449 SSL G Bus Compressor Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 450 General Usage Notes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 452 Chapter 47. Studer A800 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
TABLE OF CONTENTS 610 History . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
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 PlugIns. Documentation for the product line is available in written, video, and online formats. ReadMe The ReadMe contains information that may not be available in other locations. Please review all the information in the ReadMe before installing or using UAD Powered Plug-Ins. The ReadMe.
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.
Online Documentation The technical support pages on our website offer a wealth of helpful information that is not included in the documentation contained within the software bundle. Please visit our support pages for important technical information including the latest release notes, host application notes, and more. The main UAD Powered Plug-Ins support website is: Support Website • www.uaudio.
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.
CHAPTER 2 Ampex ATR-102 Mastering Tape Recorder It's Not a Record Until it's Mastered on an Ampex® Tape Machine. For more than three decades, the two-channel Ampex ATR-102 Mastering 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.
Ampex ATR-102 Screenshots Figure 1. The UAD Ampex ATR-102 plug-in window Figure 2.
Operational Overview Famous Tape Sound The UAD Ampex ATR-102 provides all of the original unit’s desirable analog sweetness. Like magnetic tape, users can dial in a clean sound, or just the right amount of harmonic saturation. Mixdown Tape Deck The primary purpose of the UAD Ampex ATR-102 is to obtain tape mixdown sonics within the DAW environment.
Ancillary Noises Tape recorders have inherent signal noises that are a by-product of the electro-mechanical nature of the machine. While “undesirable” tape system noise is historically considered a negative and was an attribute that pushed the technical envelope for better machine design and tape formulas (and ultimately, “noiseless” digital recorders), noise is still an ever-present characteristic of the sound of using tape and tape machines.
Low Level Tuning Even though automatic calibration is available, the individual controls that adjust calibration are exposed for sonic manipulation. Playback EQ, record (tape) EQ, and record bias can easily be altered for manual calibration and/or creative purposes. Manual Calibration Tools UAD Ampex ATR-102 includes the full suite of tools required to manually calibrate the recorder.
Primary Controls Meters The two Meters display signal levels of the plug-in for the left and right channels. Meter ballistics of the original hardware are modeled. The Meters can be switched to display input or output levels in peak or 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.
Clip LED The left and right channels each have a Clip LED, just above the Meter. The Clip LED is not in the original hardware; it is a UAD-only feature. The Clip LED illuminates only when the machine’s audio electronics clip. The Clip LED is not affected by the recorded tape signal, even if the tape is overloaded and distorting. Reproduce adjusts the signal level coming off the virtual tape before the signal is sent to the Meters. There are two Reproduce controls, one each for the left and right channels.
Record is a primary “color” control for the plug-in. Just like genuine magnetic tape, lower Record levels will have a cleaner sound, while higher levels result in more harmonic saturation and coloration. Higher Record levels will also increase the output level from the plug-in. The Reproduce control can be lowered to compensate if unity gain operation is desired.
When Link is active, automation data is written and read for the left channel only. In this case, the automation data for the left will control both channels. Additionally, changing the right channel parameters from a control surface or when in “controls only” (non-GUI) mode will have no effect. Unlink When Unlink is active, the controls for the left and right channels are independent. When unlinked, automation data is written and read by each channel separately.
OFF is similar to the Thru position in the Path Select control (page 29) except that the Meters are still active when the Thru control is used. However, in this state, the Meters indicate signal levels at the input of the plug-in prior to processing. Note: DSP usage is reduced only when DSP LoadLock is disabled. If DSP LoadLock is enabled (the default setting), activating OFF will not reduce DSP usage. Tape Speed The Tape Speed control determines the speed of the tape transport, in inches per second (IPS).
Cal Level Cal Level automatically sets tape calibration/fluxivity. The Cal Level setting takes care of the setup one would need to make under equivalent hardware operation, and sets the reference tape/flux level without disturbing the (unity) gain of the plug-in. To select the Cal Level, click the CAL button to cycle through the available levels, or click directly on the calibration value label. The active Cal Level is highlighted in yellow. The default value is +6 dB.
Table 1. Tape Manufacturer’s Recommended Calibration Levels 456 +6 dB 355 nWb/m 468 +6 dB 355 nWb/m 900 +9 dB 502 nWb/m GP9 +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 1/4,” 1/2,” or 1” can be selected.
Thru Thru is a processor bypass control. When Thru is enabled, all controls are inactive, emulation processing is disabled, and DSP usage is reduced. Thru behavior is similar to that of the OFF position in the POWER control (page 26), except that the Meters are still active in Thru mode. In this state, the Meters indicate signal levels at the input of the plug-in prior to processing. Note: DSP usage is reduced only when DSP LoadLock is disabled.
Secondary Controls The secondary controls (Figure 4 below) adjust the various calibration, ancillary noise, tone generator, and tape delay parameters. The secondary controls panel is accessed by clicking the OPEN button beneath the AMPEX label. Figure 4.
After Auto Calibration occurs, the automatically adjusted parameters can be modified to any other value if desired. If a calibration parameter is adjusted while Auto Cal is ON, the ON LED illuminates in red instead of green, indicating that the system is no longer in the calibrated state. If the moved controls are subsequently returned to their original position, the LED will return to its green state, indicating the unit is back in calibration.
Repro HF Adjusts the tape playback high frequency content when Path Select is set to Sync or Repro. Repro LF Adjusts the tape playback low frequency content when Path Select is set to Sync or Repro. Bias This control adjusts the amount of bias in the record signal. Bias is defined as an oscillator beyond the audible range applied to the audio at the record head, allowing for adjustment of the record behavior. Ideal bias voltage settings provide maximum record sensitivity and low distortion.
When Tape Speed is set to 30 IPS, the green Emphasis EQ LEDs are not illuminated (and cannot be switched), indicating that the Emphasis EQ is set to AES. However, the Hum frequency can still be set for 30 IPS mode by setting Emphasis EQ to NAB (for 60 Hz) or CCIR (for 50 Hz) prior to setting Tape Speed to 30 IPS. Note: When Tape Speed is 3.75 IPS, only 60 Hz is available. Hiss Hiss determines the amount of tape hiss in the tape playback signal.
Wow usually refers to very low frequency fluctuations, while Flutter refers to faster fluctuations. Wow and flutter is measured as the percentage of deflection from the original pitch. Both are more pronounced at lower tape speeds. Note: Wow and Flutter levels change with Tape Speed, but they are not affected by automatic calibration. Wow Determines the amount of Wow in the signal. Wow & Flutter Enable must be ON for this control to function. Flutter Determines the amount of Flutter in the signal.
Tape Delay Enable These buttons are global enable/disable controls for the Tape Delay effect. When Tape Delay is ON, its red numerical display is active, and other Tape Delay parameters can be adjusted. Dry/Wet Mix The Dry/Wet pushbuttons control the mix of the Tape Delay effect. The amount of dry and wet signals are displayed as percentages. Click the Dry button to increase the dry signal level by 1%, or the Wet button to increase the delayed signal level by 1%.
These controls are the suite of tools included to perform manual calibration of the recorder. These UAD-only tools are not in the original hardware. Manual calibration is entirely optional, as the Auto Cal feature can quickly and automatically calibrate the system.
The Manual Cal knob performs two functions: it sets the signal level of the “external” test tone generator for record calibration, and specifies when alignment tapes are to be used for playback calibration. Manual Cal Knob When set to –16 dB, –6 dB, or +4 dB, a generated sine wave test tone at the frequency specified by the Tones buttons is sent to the input of the record circuitry. This mode emulates sending external test tones into the system.
Manual Calibration Procedure Manual calibration tools are provided so expert users can calibrate the system to their preferred methods for obtaining desired results. For example, some technicians may prefer adjustments for lowest distortion at a certain frequency; setting bias for maximum sensitivity (instead of overbiasing); or other non-standard techniques.
To manually calibrate UAD Ampex ATR-102: Repro Level Calibration 1. Set the Manual Cal Knob to the “MRL” position. The built-in alignment tape tone will sound and its level can be viewed on the Meters. 2. Set the Tones frequency to 1 kHz. 3. Adjust Reproduce (output) so the Meters display 0 dB. Repro EQ Calibration 4. Set the Tones frequency to 10 kHz. 5. Adjust Repro HF (not to be confused with HF EQ) so the Meters display 0 dB. 6. Set the Tones frequency to 100 Hz. 7.
11. Increase Bias (clockwise) until the meter level is reduced by –3.5 dB from its maximum (for 3.5 dB of overbias; see Manual Calibration Notes).* *When calibrating at 3.75 or 7.5 IPS, the tone generator is at a lower level, therefore meter resolution is decreased. To increase meter precision when adjusting bias at the lower tape speeds, consider temporarily increasing the reproduce level. Record Level Calibration 12. Set the Tones frequency to 1 kHz. 13.
Manual Calibration Notes • 0 dB on the output meter represents +4 dBm (and –12 dBFS digital) when Reproduce is in its calibrated position, which is marked with the “red arrow sticker.” • For proper calibration, follow the entire calibration procedure in order. • This example uses 3.5 dB overbias. The amount of gain reduction in step 12 determines the amount of overbias. In some cases we used more than 3.5 dB of overbias to achieve a flatter response.
Parameter Dependencies Available Settings Some ATR-102 parameter value ranges depend on the value of other parameters. These dependencies are listed in Table 6 below. Table 6.
Original Ampex ATR-102 Mastering Recorder Brochure UAD Powered Plug-Ins Manual - 44 - Chapter 2: Ampex ATR-102
CHAPTER 3 API 500 Series EQ Collection Introduction The 550A and 560 modular EQs are some of the most popular and enduring mixing and tracking processors ever made. The 500 series modules from API are true industry standards, found in professional multi-channel consoles to Lunchbox racks in the humblest of project studios.
API 500 Series EQ Collection Screenshots Figure 5.
Operational Overview API 500 Series Collection The API 500 Series EQ Collection includes the UAD API 550A and UAD API 560 plug-ins, which are officially licensed from and endorsed by Automated Processes Inc. Both plug-ins meticulously model the entire electronic path, including custom API 2520 op-amps, transformers, band interactions, and internal clipped filter nonlinearities.
API 550A Controls Band Controls The three EQ bands (HF/MF/LF) are controlled by dual-concentric switches. The inner knob controls the band frequency and the outer knob controls the band gain. Available values for these controls are listed in Table 7 below. Table 7. API 550A Frequency and Gain Values Band Frequency Values Gain Values High Frequency (HF) 5, 7, 10, 12.5, 15 (kHz) Mid Frequency (MF) 0.4, 0.8, 1.
Bandpass Filter This switch (“FLTR”) applies a 50 Hz – 15 kHz bandpass filter to the entire signal. The bandpass filter is completely independent from the from the three main band filters. The HF and LF bands are normally in bell mode. When the Bell/Shelf button is engaged for the band (in the darker “down” position), the band is switched to shelving mode. Bell/Shelf Switches LF Shelf When the LF Shelf button is engaged, the low frequency band is switched to shelving mode.
API 560 Controls Note: Like the original 560 hardware, the signal is boosted by approximately 1 – 1.5 dB even when all gain sliders are set to 0 dB. Gain Sliders Each of the 10 sliders controls the gain for one frequency band. Each band can be adjusted to boost or cut the frequency by up to ±12 dB. The available band frequencies are listed in Table 8 below. Table 8.
Historical Background API (Automated Processes Inc.) was formed in 1968 with Saul Walker and Lou Lindauer. API is perhaps most noted for their modular approach to equipment manufacturing and for their now legendary 2520 amplifier. To this day, the extraordinary headroom made possible with the 2520 offers consistent analog performance even when using radical EQ curves. API quickly became the leading audio broadcast console manufacturer for radio and television networks and high profile stations.
CHAPTER 4 API Vision Console Channel Strip Introduction The API Vision Console Channel Strip plug-in for the UAD platform is based on API’s flagship console found in studios and sound stages across the globe. The plug-in includes five indispensable modern production API modules available in the Vision console: The custom 2520 op-amp based 212L Preamp, 215L Sweep Filters, 550L EQ, 225L Compressor/Limiter, and the 235L Gate/Expander.
The 235L Noise Gate/Expander is one of the fastest noise gates available. The API 235L can reduce noise in any type of program without losing any part of the source. Its extreme flexibility and superb sound make it ideal for all recording or mixing studio applications. The Expander function uses a 1:2 ratio, allowing the signal to “sneak up” to the full signal level without any loss of “under threshold” vocal or percussion nuances.
Operational Overview Modular Design Like the original hardware, the API Vision Console Channel Strip plug-in has a modular design. Each module controls a different signal processing function, and associated controls are grouped within each module.
Displayed Values Knob settings, when compared to the graphical user interface silkscreen numbers, may not match the actual parameter values. For example, in the 215L Sweep Filters module, the highest value shown in the plug-in window is 20 kHz. However, the actual value when the knob is at maximum is 40 kHz. This behavior is identical to the original hardware, which is modeled exactly.
Hardware Input Impedance All Apollo mic preamps feature input impedance switching in analog hardware that can be physically switched by Unison plug-ins for physical, microphone-to-preamp resistive interaction. This impedance switching enables Apollo’s preamps to physically match the emulated unit’s input impedance, which can significantly impact the sound of a microphone. Because the electrical loading occurs on input, prior to A/D conversion, the realism is faithful to the original target hardware preamp.
API Vision Console Channel Strip Controls 212L Microphone Preamplifier 212L Gain This knob adjusts the amount of gain applied to the input signal. The available range is 30 dB to 65 dB. The default value is 40.5 dB. 212L Pad When enabled, the input signal level is attenuated (lowered) by 20 dB. The pad is engaged when the red indicator is lit. Pad can be used to reduce signal levels when undesirable overload distortion is present at low preamp gain levels.
215L High/Low Sweep Filters The 215L offers two sweepable cut filters, one each for low and high frequencies. The original hardware is transformer coupled and uses a passive filter circuit design for smooth tone. 215L Lo-Pass The Lo-Pass (high cut) filter has a continuous range of 643 Hz to 40.8 kHz. The slope of this filter is 6 dB per octave. The default value is 40 kHz. 215L Hi-Pass The Ho-Pass (low cut) filter has a continuous range of 12 Hz to 596 Hz. The slope of this filter is 12 dB per octave.
235L Gate/Expander The 235L Gate/Expander module operates in either gate or expansion mode. Two attack speeds and a continuously variable release time are available in both modes. 235L Threshold Threshold defines the input level at which expansion or gating occurs. The available range is from +25 dB to -45 dB. The default value is 0 dB. Signals below the threshold level are processed by the module. Signals above the threshold are unaffected.
Release When the input signal drops below the threshold level and the Release/Hold switch is set to Release, this knob sets the amount of time it takes for signals to decay to the Depth level. 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 percussion and other instruments with very fast decays.
Expansion allows the signal to “sneak up” to the full signal level without any loss of “under threshold” nuances. 235L Meter This meter displays, in dB, the amount of gain attenuation (downward expansion) occurring in the 235L module. 235L On This button enables the 235L module. The module is active when the button’s green indicator is lit. Note: UAD DSP load is reduced when this module is inactive (unless DSP LoadLock is enabled).
225L Ratio Ratio defines the amount of gain reduction applied to signals above the threshold. For example, a value of 2 (expressed as a 2:1 ratio) reduces the signal level above the threshold by half, with an input signal level of 20 dB being reduced to 10 dB. A value of 1 yields no gain reduction. When the control is at maximum (∞), the ratio is effectively infinity to one, yielding the limiting effect. The available range is 1:1 to infinity. The default value is 4:1.
225L Type The Type control switches the 225L compressor’s control side chain signal to use either a feed-back (OLD) or feed-forward (NEW) design, providing two types of gain reduction. The default value is Old. Compressors typically have a side chain control signal based on either feedback or feed-forward designs. NEW feed-forward gain reduction is typical of newer VCA type compressors that rely on RMS detectors for the side chain circuit.
550L Four-Band Equalizer The 550L EQ is divided into four frequency bands: High Frequency (HF), High Midrange Frequency (HMF), Low Midrange Frequency (LMF), and Low Frequency (LF). The 550L features API’s “Proportional Q” which continuously narrows the bandwidth of the filter as band gain is increased, providing (as stated by API) “an uncomplicated way to generate acoustically superior equalization.” The boost and cut characteristics are identical, allowing previous actions to be undone if desired.
Gain The gain for the band can be set using any of these four methods: • Drag the outer concentric knob handle to the desired value • Click the “+” or “-” text labels to increment/decrement values • Hover over the outer concentric knob then use the mouse scroll wheel • Click directly on the gain value label to switch to that value (this method works only when Controls Mode is set to “Circular” in the Configuration panel of the UAD Meter & Control Panel application) Peak/Shelf Switches The HF and LF bands
Figure 11. The 550L always precedes the side chain tap when PREDYN is active 550L SC (Dynamics Side Chain) This control enables the side chain function for the 550L EQ. When the 550L side chain is active, signal output from the 550L module is removed from the audio path, and is instead routed to control the 235L and 225L dynamics modules. The default value is Off. Figure 12.
550L EQ This button enables the 550L module. The module is active when the button’s green indicator is lit. Note: UAD DSP load is reduced when this module is inactive (unless DSP LoadLock is enabled). Global Module Output Meter The vertical LED-style metering provides a visual indication of relative signal peak levels at the output of the plugin.
Power The plug-in is active when the POWER switch is engaged and its associated LED is lit. When this button is off, all plug-in processing is disabled and UAD DSP usage is reduced (unless DSP LoadLock is enabled). Historical Background API (Automated Processes Inc.) was formed in 1968 with Saul Walker and Lou Lindauer. API is perhaps most noted for their modular approach to equipment manufacturing and for their now legendary 2520 amplifier.
CHAPTER 5 Cambridge EQ Overview The UAD Cambridge EQ plug-in is a mastering-quality, no-compromise equalizer 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 handling 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 Display The Response Curve Display plots the frequency response of the current Cambridge EQ settings. It provides instant visual feedback of how audio is being processed by the equalizer. Figure 15. Cambridge EQ Response Curve display The entire audio spectrum from 20 Hz to 20 kHz is displayed along the horizontal axis.
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. Four vertical ranges can be selected with the Zoom buttons: ±5, ±10, ±20, and ±40 dB. Figure 16.
Master Level Knob This control adjusts the signal output level of Cambridge EQ. This may be necessary if the signal is dramatically boosted or reduced by the EQ settings. The available range is ±20 dB. A/B Selector Button The A/B Selector switches between two separate sets of Cambridge EQ plug-in values. This feature enables easy switching between two completely independent EQ curves which can be useful for comparison purposes or for automating radical timbre changes.
Low Cut / High Cut Filters The Low Cut and High Cut filters are offered in addition to the five parametric/shelf bands. A wide range of filter types is provided to facilitate tonal creativity. Many filters that are available are represented. Three controls are offered: Cut Type, Enable, and Frequency. Each control is detailed below. The Cut Type menu determines the sound of the low and high cut filters. To view the Cut Type menu, click and hold the green cut type button.
Cut Frequency Knob This knob determines the cutoff frequency for the Cut filters. The available 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. Each band has identical controls, the only difference is the frequency range values. The function of the controls is similar in both parametric and shelf modes.
Table 10. 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 frequency will be limited. Note: Gain Knob This parameter determines the amount by which the frequency setting for the band is boosted or attenuated. The available range is ±20 dB.
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. Type I When set to Type I, the bandwidth remains at a fixed Q regardless of the gain setting for the band; there is no Q/Gain interdependency. In addition, there is a finer resolution of the Q knob in the middle of its range. This makes it easier to achieve subtle bandwidth changes. Note that the Q value and knob positions do not change as the gain is modified.
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 20. Figure 20. Parametric Type II response Type III When set to Type III, there is a Q/Gain dependency on boost and attenuation. The bandwidth increases continuously as the gain is boosted and attenuated. The Q knob position determines the maximum Q at full gain.
Shelf EQ Each band can be switched from parametric mode to shelf mode by clicking the shelf enable button. The button is off by default. To enable shelving on any band, click the shelf button. Shelf Enable 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 23 on page 79) when shelf mode is active.
Figure 22. Shelf Type A Shelf Mode Indicator Line Figure 23. Shelf Type B Figure 24.
CHAPTER 6 Cooper Time Cube Dual Mechanical Delay Line The original Cooper Time Cube was a Duane H. Cooper and Bill Putnam collaborative 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. The Cooper Time Cube is famous for its spectacular short delay and doubling effects and its uncanny ability to always sit perfectly in the mix.
Design Overview The original UREI/Universal Audio Model 920-16 Cooper Time Cube hardware (see “Cooper Time Cube Hardware” on page 85) has two audio channels, 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. The coils for each channel are at fixed but different lengths, which define the available single delay times of 16ms for channel A and 14ms for channel B.
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. Full processor bandwidth is obtained with the knob in the fully counter-clockwise position. Echo A/B These two “windows” display the current delay times of channels A and B.
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 Decay ranges. Unlike the other parameters, the A and B labels for Color are for reference only. They do not represent the left and right channels. Note: Color can be subtle, and its affect can vary depending on the value of Coils and/or Decay. Treble Treble controls the high frequency response in the delayed portion of the signals.
Channel Controls The channel controls affect each channel of the processor independently. The control functionality is identical for each channel. “A” indicates the left channel and “B” is the right channel. Figure 27. The channel controls Delay A/B Delay controls the delay time for each channel of the processor. The selected value is shown in the Echo display (“Echo A/B” on page 82). The available delay range for each channel is 5 milliseconds to 2.5 seconds (2500ms).
Echo Volume A/B 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” label text to mute/unmute the delayed output. Cooper Time Cube Hardware Figure 28. The original Cooper Time Cube hardware front panel Figure 29.
CHAPTER 7 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. Individual effects in the CS-1 Channel Strip can be bypassed when not in use to preserve UAD DSP use. The CS-1 effects can also be accessed individually by using the individual plug-ins.
EX-1 Equalizer and Compressor Figure 31. 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. Similarly, the last two bands can be enabled to function as either a high-shelf or low-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. Frequency (fc) Knob Determines the center frequency to be boosted or attenuated by the Gain setting. The available range is 20 Hertz to 20 kiloHertz. When operating at sample rates less than 44.1kHz, the maximum frequency will be limited. Bandwidth (Q) Knob Sets the proportion of frequencies surrounding the center frequency to be affected.
Ratio Knob Determines the amount of gain reduction used by the compression. For example, 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 compression. Values beyond 10 yield a limiting effect. The range is 1 to Infinity. Threshold Knob Sets the threshold level for the compression. Any signals that exceed this level are compressed. Signals below the level are unaffected.
DM-1 Delay Modulator Figure 32. 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. L-Delay Knob Sets the delay time between the original signal and the delayed signal for the left channel. When the Mode is set to one of the delay settings, the maximum delay is 300 msec.
Mode Pop-up Menu Determines the DM-1 effect mode. The available modes are: Chorus, Chorus180, QuadChorus, Flanger1, Flanger2, Dual Delay, and Ping Pong Delay. In addition to reconfiguring the DM-1’s settings, the Mode also determines the available parameter ranges for L/R Delay and Depth. In Chorus mode, both oscillators (or modulating signals) are in phase. In Chorus 180 mode, both oscillators (the modulating signals) are180 degrees out of phase (inverted).
The RECIR units are expressed as a percentage in all Modes except Dual Delay and Ping Pong. In these modes, RECIR values are expressed as T60 time, or the time before the signal drops 60 decibels. Damping Knob This low pass filter reduces the amount of high frequencies in the signal. Turn down this control to reduce the brightness. Higher values yield a brighter signal. Damping also mimics air absorption, or high frequency rolloff inherent in tape-based delay systems.
Link Button This button links the left and right delay knobs so that when you move one delay knob, the other follows. The ratio between the two knobs is maintained. Figure 33. The DM-1L includes a Link button RS-1 Reflection Engine Figure 34. The RS-1 Reflection Engine plug-in window Overview The RS-1 Reflection Engine simulates a wide range of room shapes, and sizes, to drastically alter the pattern of reflections.
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. Shape Pop-up Menu Determines the shape of the reverberant space, and the resulting reflective patterns. Table 11.
Recirculation allows both positive and negative values. The polarity refers to the phase of the delays as compared to the original signal. If Recirculation displays a positive value, all the delays will be in phase with the source. If it displays a negative value, then the phase of the delays flips back and forth between in phase and out of phase. Damping Knob This low pass filter reduces the amount of high frequencies in the signal. Turn down this control to reduce the brightness.
CHAPTER 8 dbx 160 Compressor/Limiter Overview The dbx® 160 Compressor/Limiter is an officially licensed and faithful emulation 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 Threshold are unaffected. The available range is from –55 dB to 0 dB. The numbers on the graphical interface indicate volts, as on the original hardware.
inside the T4 module. Use the LA-2 with legato tempos and your most vowellike sources for a transparency and sublime mood unlike any other compressor.” on page 470 for more information about compressor/limiter theory of operation. Output Gain controls the signal level that is output from the plug-in. The available range is ±20 dB. Output Gain Generally speaking, adjust the Output control after the desired amount of compression is achieved with the Threshold and Compression controls.
CHAPTER 9 DreamVerb Overview DreamVerb™, Universal Audio’s unique 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 36. The DreamVerb plug-in window Signal Flow Figure 37 illustrates the signal flow for DreamVerb. The input signal is equalized then delay lines are applied to the early reflection and late field generators. The resulting direct path, early reflection, and late-field reverberation are then independently positioned in the soundfield. Pan Direct Path Source Input Wet/Dry Mix EQ Delay Early Reflections Gain & Mute Pans & Distance Delay Gain Output LateField Reverb Figure 37.
The DreamVerb user interface (Figure 36 on page 100) is similarly organized. 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 40 on page 103). Early reflection pre-delay, slope, timing, and amplitude are specified in the Reflections panel (Figure 42 on page 108).
Bypass switch Band Amplitude control bats Band 1 (low shelving) control Band 2, 3, and 4 Edge control bats Band 5 (high shelving) control Figure 38. DreamVerb Resonance panel Bypass The equalizer can be disabled with this switch. When the switch is off (black instead of grey), the other resonance controls have no effect. This switch has no effect on the direct signal path. Band Amplitude Each of the five bands has its own amplitude (gain) control.
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 rightmost horizontal line (the ones without control bats) up or down, which boosts or cuts the energy at these frequencies. Drag these control handles up or down for shelving EQ. Figure 39.
DreamVerb lets you specify two room shapes that can be blended to create a hybrid of early reflection patterns. The first and second shape each have their own menu. The available shapes are the same for each of the two shape menus. Shape Menus The first shape is displayed in the upper area of the Shape panel, and the second shape is displayed in the lower area. To select a first or second shape, click its shape pop-up selector menu to view the available shapes, then drag to the desired shape and release.
Materials Panel The parameters in the Materials panel, in conjunction with the Shape panel (Figure 40 on page 103) and Reverberation panel (Figure 43 on page 109) effect the spatial characteristics of the reverb. The material composition of an acoustical space effects how different frequency components decay over time. Materials are characterized by their absorption rates as a function of frequency—the more the material absorbs a certain frequency, the faster that frequency decays.
Materials Menus DreamVerb lets you specify two room materials, which can be blended to create a hybrid of absorption and reflection 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.
Materials Blending Bars The Materials Blending Bars (see Figure 41 on page 105) are used to blend the 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. Drag the bar to the right to emphasize the first material; drag to the left to emphasize the second material.
ER End control bat (time & amplitude) Bypass switch Materials Filtering control bat ER Start control bat (predelay & amplitude) Late-field relative timing display Figure 42. 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. This switch has no effect on the direct signal path. Reflections Start This bat controls two early reflections start parameters.
Late-Field Relative Timing To highlight the relative timing relationship between the early reflections and 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. Reverberation Panel The Reverberation panel (Figure 43) contains the parameters that control the late-field (LF) reverb tail for DreamVerb.
Late-Field Start This parameter defines when the late-field reverb tail begins (the delay between the dry signal and the onset of the LF) in relation to the dry signal. Amplitude & Slope This bat controls two late-field parameters. Dragging the bat vertically controls the maximum amplitude of the LF reverb energy. Dragging it horizontally controls the LF slope (fade-in) time. Decay Time This control effects the length of the reverb tail.
Figure 44. 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. A value of <0> places the signal in the center of the stereo field. Note: If the DreamVerb “Mix” parameter (page 112) is set to 100% wet or the Wet button is active, these sliders have no effect.
Distance DreamVerb allows you to control the distance of the perceived source with this slider. In reverberant environments, sounds originating close to the listener have a different mix of direct and reflected energy than those originating further from the listener. Larger percentages yield a source that is farther away from the listener. A value of 0% places the source as close as possible to 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. Mix The wet and dry mix of DreamVerb is controlled with this slider. The two buttons above this slider labeled “D” and “W” represent Dry and Wet; clicking either will create a 100% dry or 100% wet mix.
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 preset 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” in Chapter 7 of the UAD System Manual). Preset Design Tips Here are some practical tips for creating useful reverbs with DreamVerb.
• 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/mastering presets, use the EQ to cut most of all LF input, which yields added ambience without mucking up the mix. This is a powerful EQ, so experiment! • Try different diffusion settings for your preset (the slider on the right of the Reverberation panel). Diffusion radically alters the reverberation sound and is source dependent.
CHAPTER 10 Empirical Labs EL7 FATSO Introduction FATSO Jr. Endorsed and scrutinized for accuracy by designer Dave Derr of Empirical Labs (originator of the hugely popular Distressor), UA has painstakingly recreated the highly regarded FATSO Jr. as a plug-in, capturing the sonic nuances of the hardware.
FATSO Screenshots Figure 46. The FATSO Jr. plug-in window Figure 47. The FATSO Sr. plug-in window FATSO Functional Overview Four Processing Types The FATSO was essentially designed to integrate frequencies in a musical manner and provide some foolproof vintage sounding compression. Generally, it is difficult to make the unit sound unnatural due to its vintage topology. FATSO provides four types of processing.
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 harmonic is considered to be the warmest and most “consonant” harmonic distortion. Warmth Processor High Frequency Saturation This circuit is meant to simulate the softening of the high frequencies that occurs with analog tape. Basically, as the Warmth is increased, overly bright signals and transients will be quickly attenuated.
Transformer design and use is an art, and there are always trade-offs. However, 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 selectable 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 during mixdown. Spank Spank mode (red LED) is a radical limiter type compressor that was specifically designed to emulate the nice squeeze of the older SSL talkback compressors from the 70's & 80's, but with quite a bit of higher fidelity. Note that Spank's aggressive nature will tend to dominate when combined with any of the other modes.
Channel Controls 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. Input When the compressor is active (see “Compressor Mode” below), higher input values also result in more compression, as indicated by the gain reduction meters (page 122).
Table 12. Compressor Mode LED States GR Meter 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) Red Spank Red + Green Spank + Buss Red + Yellow Spank + General Purpose Red + Green + Yellow Spank + Tracking The Gain Reduction Meter displays the amount of gain reduction occurring within the FATSO compressor, expressed as negative dB values.
This black button is a multifunction control. Clicking the button repeatedly cycles through Tranny, Bypass, and Tranny Off modes. The currently active mode is indicated by the adjacent LED's. Bypass/Tranny Tranny (green LED) The Tranny processor is active in this mode (see “The Tranny Processor” on page 118 for a detailed description of this mode). The Tranny circuit adds frequency “rounding,” low order clipping, intermodular distortion and transient clipping. On FATSO Sr.
Global Controls The global controls are not channel-specific; they apply to both channels. The control signal sidechains of the gain reduction processors for channels 1 and 2 can be linked using the Link Compress function. Link Compress To activate Link Compress, click the LINK COMPRESS text or LED on Ch1, on the left. The feature is active when the LED is illuminated. In typical use on stereo signals, Link Compress should be active so the stereo imaging is maintained.
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.
Filter regulates the cutoff frequency of the filter on the compressor'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. When the compressor is disabled, Filter has no effect and its LED turns off. When the compressor is enabled, Filter returns to its original value.
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” on page 121). Release Release sets the amount of time it takes for compression to cease once the input signal drops below the threshold level.
This control determines the amount of Tranny processing (see “The Tranny Processor” on page 118 for a detailed description). Higher values make the Tranny effect more prominent. Increasing the Tranny level also increases the signal THD (see “THD Indicators” on page 121), and the sensitivity of the Warmth processor (page 122). A value of 5 is the unity setting. Tranny Level Note: This control has no effect when the Tranny processor is inactive (see “Bypass/Tranny” on page 123).
CHAPTER 11 EMT 140 Plate Reverb Overview EMT’s founder Wilhelm Franz made a breakthrough in 1957 with the release of the EMT 140, which utilized a resonating metal plate to create ambience. Nothing is quite like the wonderfully lush and distinctive tone of plate reverb that still endures as part of the fabric of modern music.
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. Note: When adjusting parameters, keyboard shortcuts are available for fine, coarse, and other control methods. See “Shortcuts” in Chapter 7 of the UAD System Manual for details.
Reverb Controls Plate reverb systems are extremely simple: A remote damper setting, and a high pass or shelf filter found at the plate itself. Additional manipulation is often used, including reverb return equalization, which is typically achieved at the console. Predelay is/was often achieved when necessary with tape delay, sending the return to a tape deck. Different tape speeds allowed different pre-delay amount.
Stereo Controls 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. Width Balance This control balances the level between the left and right channels of the reverb return. Rotating the knob to the left attenuates the right channel, and vice versa (it is not a mono pan control).
Because this is a shelving EQ, all frequencies below this setting will be affected by the low band Gain value. Low Gain This parameter determines the amount by which the transition frequency setting 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). High Frequency This parameter determines the high shelving band transition frequency to be boosted or attenuated by the high band Gain setting.
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. Output Meter Blend Controls Predelay The amount of time between the dry signal and the onset of the reverb is controlled with this knob. The range is 0.0 to 250 milliseconds. This control uses a logarithmic scale to provide increased resolution when selecting lower values. When the knob is in the 12 o’clock position, the value is 50 milliseconds.
This toggle switch enables or disables EMT 140. You can use it to compare the processed settings to the original signal, or to bypass 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. Power Switch Note: The EMT 140 distills 1800+ pounds of classic vintage reverb into a single plug-in. Exercise caution when lifting.
CHAPTER 12 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 indispensable tool within the record-making elite and is widely considered one of the best-sounding reverbs ever made.
EMT 250 Screenshot Figure 49. 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. Each program mode has up to five parameters that can be modified by the four main control “levers” plus the front/rear switch. The function of these controls varies per program mode (see below).
Each unique parameter in the plug-in retains a distinct value, but only the parameters that are active in the current program mode are visible in the graphical 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. Important: The value of lever parameters that are not active in the current program mode are not saved in sessions or presets.
In some program modes, the yellow “LED ring” around the control is illuminated 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. Automation Some EMT 250 control functions change depending on the active mode (see “Variable Control Functions” on page 137 and Table 13 below).
Program Mode Controls The details of each unique program mode are below, followed by descriptions of the global controls, which affect all program modes. Control Functions Table 13 displays the parameter that each control is mapped to for each of the EMT 250 program modes. See “Variable Control Functions” on page 137 for details. Table 13.
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. The decay time range (at 1 kHz) is 0.4s to 4.5s, selectable via 16 steps. LF Decay (Lever 2) Lever 2 controls the low frequency decay time (at 300 Hz).
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 obtained by setting the coarse, fine, and predelay times to their respective maximum values.
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 control surfaces or automation, nor is it saved in sessions or presets. Predelay (Lever 4) Lever 4 can be used as a common predelay to both channels (the predelay time is added to the delay times of both channels). See “Lever 4 Predelay” on page 138 for more information.
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. Instead, 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. See “Lever 4 Predelay” on page 138 for more information. Note: Levers 2 and 3 have no effect in Phase program mode. Front/Rear The Front/Rear Outputs switch is illuminated in Phase program mode.
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” combines both the Left Front and Left Rear on the left side, and Right Front and Right Rear on the right. “IV” combines Left Front, Left Rear and Right Rear on the left side, and Left Rear, Right Front and a phase inverted Right Rear on the right.
Note: Levers 1 and 2 both control the echo time, but these parameters are not individually exposed for external control surfaces and automation. Instead, 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. The red LEDs on left side of lever 3 display the current value; the green LEDs on the right side of lever 3 are inactive. Four multipliers are available: x 0.
Predelay (Lever 4) Lever 4 is used as a typical reverb predelay parameter. See “Lever 4 Predelay” on page 138 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” on page 138 for more information. Global Controls The global controls are not program-specific; they apply to all program modes. The Power button (the red EMT logo) determines whether the plug-in is active.
The Dry/Wet slider control determines the balance between the original and the processed signal. The range is from 0% (dry, unprocessed) to 100% (wet, processed signal only). Dry/Wet This control uses a logarithmic scale to provide increased resolution when selecting lower values. When the slider is in the center position, the value is 15%. Note: If Wet Solo is active, adjusting Dry/Wet will have no effect. Wet Solo The Wet Solo button puts the EMT 250 into “100% Wet” mode.
CHAPTER 13 EP-34 Classic Tape Echo EP-34 Overview The EP-34 combines EP-3 and EP-4 sonics and features to achieve the best of the later solid-state Echoplex* designs. The Echoplex uses an infinite tape loop combined with a sliding record head that allows the user to achieve the desired delay length.
EP-34 Tape Echo Screenshot Figure 50. 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 151). The parameter can be adjusted by using the metallic “slider handle” or the “slider nose” (both sliders control the same parameter; see Figure 51). Figure 51.
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). See Chapter 8 in the UAD System Manual for more information about tempo synchronization.
This knob determines the wet/dry mix of the delayed signal. In the minimum position, the “dry” signal is colored by the circuitry of the modeled emulation. Rotate the control clockwise for louder echo. Reducing the control to its minimum value will mute the delay. Echo Volume The EP-34 models the unusual taper of this control that is found on the original hardware. It is normal operation to have the control in the 85–95% range to get a “50/50” wet/dry balance.
Pan sets the position of the delayed (wet) signal in the stereo field; it does not affect the unprocessed (dry) signal. Echo Pan 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. Input The original hardware unit had two inputs: Instrument and Microphone. The Input switch on the EP-34 toggles between the gain levels of these two inputs.
Sync This switch engages Sync mode for the plug-in. In Sync mode, delay times are synchronized to (and therefore dependent upon) the master tempo of the host application. When Sync is toggled, parameter units are converted between milliseconds and beats to the closest matching value. See Chapter 8 in the UAD System Manual for more information about tempo synchronization. Wet The Wet switch puts the EP-34 into “100% Wet” mode. When Wet is on, it mutes the dry unprocessed signal.
The EP-3 is the favored unit by guitarists, and EP-4 is the last unit that was released and has an improved feature set over its predecessors such as metering and tone controls making it even more useful as a mix tool. Some didn’t like the EP-4 because of a noise reduction circuit that was added that was not implemented correctly.
CHAPTER 14 Fairchild Tube Limiter Collection The Gold Standard in Vintage Tube Limiters In the annals of compressor history, the products produced by Fairchild are some of the best built and most highly prized on the vintage market. The most famous Fairchild products produced were the 660 and 670 limiters, which are famous for their fantastic sound quality.
Fairchild Tube Limiter Collection Screenshots Figure 52. The Fairchild 660 plug-in window Figure 53.
Figure 54. The Fairchild 670 Legacy plug-in window Fairchild Plug-In Family The complete Fairchild family is comprised of three individual plug-ins, as seen in the screenshots above and described below. Each variation has its own unique sonic characteristics. Fairchild Tube Limiter Collection The Fairchild Tube Limiter Collection, introduced in UAD v7.4, includes two plug-ins: Fairchild 660 and Fairchild 670. Fairchild 660 The original Fairchild 660 hardware is a single-channel monophonic processor.
Fairchild 670 The Fairchild 670 is the two-channel stereo workhorse revered by engineers and producers worldwide. Second-Generation Algorithms The newer state-of-the-art algorithms in the Fairchild Tube Limiter Collection take full advantage of the extra processing power available on UAD-2 devices, along with the design sophistication and expertise gained since the original Fairchild Legacy plug-in was developed.
Fairchild 670 Legacy The original version of the Fairchild 670 was released in January of 2004 for the UAD-1 platform. To accommodate the limited DSP resources of the firstgeneration UAD-1 hardware, the transformer and I/O distortion characteristics were not modeled in this version of the plug-in.
Lateral/Vertical One of the design goals of the Fairchild 670 hardware was to facilitate its use as a limiter when producing vinyl phonograph masters. The terms lateral (side-to-side) and vertical (up-and-down) refer to the mechanical modulations in a vinyl record groove that are transduced into electrical audio signals by the phonograph stylus and cartridge. The Fairchild 670 can perform dynamics processing on the lateral (“Lat”) and vertical (“Vert”) components of stereo signals independently.
Fairchild 670 Modes 2 Compressors, 4 Modes There are two independent compressors within the Fairchild 670. Depending on the state of the AGC Switch and the Sidechain Link switch, four operating modes are possible. The modes are detailed in this section. Modes Table The switch positions required for each operating mode is shown in Table 14 below. See “Lateral/Vertical” on page 161 for an overview of these terms. Table 14.
Stereo Lat/Vert Stereo lateral/vertical (mid/side) mode, like stereo left/right mode, causes the two compressors to be linked together so that they always compress the same amount. In this mode however, the inputs to the two compressors are fed with the middle and side components of the signal respectively. This generally means that a transient which occurs in both channels will cause a bit more compression than a transient which only appears on left or right.
Meter Switch This switch determines what is displayed on the VU meters. Input, output, or gain reduction (“GR”) levels can be selected. The default value is GR. If GR is selected, the meter will show gain reduction in dB for the corresponding compressor channel. If the AGC Switch is set to left/right, the GR shown will be for the left or right channel. If the AGC switch is set to Lat/Vert, the GR shown will be for the middle (upper meter) or side (lower meter) channel.
Input Gain versus Threshold The amount of signal compression is determined by both the Input Gain and Threshold controls. If one is increased and the other decreased, the compression characteristics won’t change much, but the distortion characteristics will. The input control is located ahead of the tubes, directly “behind” the input transformer.
This switch determines whether the two compression channels will receive left/right or lateral/vertical (mid/side) signals as the inputs. AGC Switch In conjunction with the Sidechain Link switch, this control determines the operating mode of the Fairchild 670. See “Fairchild 670 Modes” on page 162 and Table 14 on page 162 for detailed mode descriptions. Note: This control is unavailable on the Fairchild 660.
Headroom Overview The Fairchild hardware units can accept an analog signal level of approximately +27 dBm before undesirable signal clipping occurs. As the signal increases up to this point however, desirable audio-path nonlinearities and “good” harmonic distortion characteristics occur. This musically pleasing “warmth” at higher levels is what gives the unit much of its revered sonic character.
Keep in mind there are no hard and fast headroom rules. Feel free to experiment with the various positions of the HR control regardless of the audio source. If it sounds good, use it! Note: The HR control does not exist on the Fairchild 670 Legacy. On the hardware unit, the Zero screw (as displayed in the Fairchild 670 Legacy) adjusted the meter pointer to compensate voltage fluctuation and component wear. Balance Balance controls the bias current balance.
DC Threshold DC Threshold 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. It’s more technically accurate to say that this control simply changes the knee width, since no matter where it’s set, the ratio always approaches true limiting eventually.
Mix The output balance between signal processed by the plug-in and the original dry source signal can be adjusted with the Mix control. Mix facilitates parallel compression techniques without having to create additional routings in the DAW. When set to 0%, only the unprocessed (dry) source signal is output. When set to 100% (the default value), only the processed (wet) signal is output. When set to 50%, an equal blend of both the dry and wet signals is output.
Historical Background The origins of the Fairchild 660/670 design come from Estonian-born immigrant Rein Narma. Les Paul hired Narma to modify his first 8-track Ampex machine. Later, Narma built consoles for Olmsted Recording, Rudy Van Gelder, and Les Paul, who then asked him to build an all-new, sonically reliable audio limiter. In the post-war years, this refugee from Soviet Russia worked for the U.S.
The Fairchild 660 and 670 Original Hardware UAD Powered Plug-Ins Manual - 172 - Chapter 14: Fairchild Tube Limiter Collection
CHAPTER 15 Harrison 32C EQ Overview The Harrison 32C is the EQ channel module from the prestigious Harrison 4032 console. Countless hit records have been made with Harrison consoles, with artists from Abba to Sade. Most notably, the 4032 is famous as the mixer from which many Michael Jackson records including Thriller—the best-selling album of all time—were made.
Harrison 32C EQ and Harrison 32C SE Controls Note: Knob settings, when compared to the graphical user interface silkscreen numbers, may not match the actual parameter values (e.g., if a knob is pointing to 8 kHz, the actual frequency may not be 8 kHz). 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. The Power switch determines whether the plug-in is active.
Low Pass (high cut) This control determines the cutoff frequency for the low pass filter. The available range is 1.6 kHz to 20 kHz. Each of the four EQ bands have similar controls. The band center frequency is controlled the top row of knobs, and the band gain is controlled by the bottom row. Four EQ Bands Low Peak The low EQ band can be operated in either peak or shelf mode. When the Low Peak switch is in the “out” position, the low EQ band operates in shelf mode.
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 output from the plug-in. The default value is 0 dB. The available range is ±10 dB. Gain Harrison 32C SE Figure 56. The Harrison 32C SE plug-in window Overview The UAD Harrison 32C SE is derived from the UAD Harrison 32C.
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” in the UAD System Manual for more information. The Harrison 32C SE does not require additional latency compensation because it is not upsampled.
CHAPTER 16 Helios Type 69 Equalizer Overview Helios consoles were used to record and mix some of the finest rock, pop and reggae classics ever produced. The Beatles, Led Zeppelin, The Rolling Stones, The Who, Roxy Music, Queen, Jimi Hendrix and Bob Marley are just a few that recorded with these amazing wrap-around consoles. Moreover, many great musicians of the era purchased Helios consoles for their personal use.
Helios Type 69 Controls The simple yet powerful Helios Type 69 Passive EQ adds a unique sonic texture 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. The Treble band is a fixed 10 kHz shelf EQ, while the Bass band functions as a stepped 50 Hz shelf filter (-3,-6,-9,-12,-15 dB) or frequency selectable Peak EQ (60, 100, 200, 300 Hz).
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. Bass 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. In this mode, the amount of gain (“bass boost”) applied to the specified frequency is determined by the Bass Gain knob.
Whether gain or attenuation is applied is determined by the Mid Type control. Note: Mid Type Mid Type specifies whether the midrange band is in Peak or Trough mode. When switched to Peak, the Mid Gain control will boost the midrange. When switched to Trough, Mid Gain will cut the midrange. Note: When using Trough, a 1 dB loss occurs on the overall output of the plug- in. This is normal; the behavior is the same in the original hardware.
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. 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.
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 - 183 - Chapter 16: Helios Type 69 Equalizer
CHAPTER 17- 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 Leveling Amplifier, the solid-state LA-3A offered a new sound in optical gain reduction, with faster attack and release characteristics that were noticeably different from its predecessor. Immediately embraced as a studio workhorse, the LA-3A is still widely used today.
LA-3A Controls Background For detailed information about compressors, see “The LA-2 captures one of the earliest Teletronix examples. This exceedingly rare unit preceded the LA2A by a few years and incredibly, still has the original T4A fully intact. The LA2 provides the slowest response and a unique “mellowed” sound due to 50 years of luminescent panel aging inside the T4 module.
CHAPTER 18 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 famously lush reverb tail — almost single-handedly defined the sound of an entire era.
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 under the Program selection. Lexicon’s distinctive Bass/Mid “split decay” adjustments and Crossover control set the highly tunable reverb image, along with the Treble Decay for rolling off high frequencies.
Lexicon 224 Screenshot Figure 60.
Operational Overview Graphical User Interface The original Lexicon 224 consists of two hardware elements. The “mainframe” 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” functions a tricky procedure.
Lexicon 224 Buttons Like the original hardware, UAD Lexicon 224 buttons are momentary-style 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; the value is actually changed only with subsequent clicks.
When configured as mono-in/mono-out (“MIMO”), output A is used exclusively except in programs 2, 4, and 9, where outputs A and C are summed into one monophonic signal. This implementation is recommended in the original hardware manual. If Rear Outs is enabled in MIMO mode, outputs B and D are used instead of A and C. See Table 17 on page 205 for a list of outputs used with each program in this configuration.
If 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; the actual decay times may not match the displayed value. Value LED The Value LED shows the units of the numerical value being displayed for a particular control.
Primary Controls Program The Program buttons (Figure 62) are used to specify which of the nine default Lexicon programs, and its associated algorithm, is active. See “Lexicon 224 Programs” on page 189 for an overview. Eight reverb programs and one chorus program are available. Click a reverb program button 1 – 8 to select that program. To select the chorus program, shift+click any program button, or click the CLK=CHORUS text label.
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 seconds to 70 seconds. This control works in conjunction with the Crossover parameter, which defines the range of the bass frequencies affected by the Bass control.
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 Decay is set very low, then adjusting Bass, Mid, and Crossover may have little to no audible effect. The available range is 100 Hz to 10.9 kHz. Tip: Treble Decay adjusts the AMOUNT of reverb tail highs, while Mid adjusts the TIME.
Higher Diffusion values are frequently desirable when the material has a lot of percussion. Higher Diffusion can also contribute to a smoother-sounding reverb. 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. Low Diffusion is also useful in classical music or in adding a sense of depth to an overall mix. Note that in Lexicon 224, lower frequencies are generally less diffuse.
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. System Noise This UAD-only control enables or disables the modeled inherent dynamic system noise of the original Lexicon 224 hardware. Disabling System Noise enables a more modern-sounding (i.e., cleaner) 224.
Mode Enhancement Mode Enhancement makes the sound of the Lexicon 224 programs more natural by preventing room modes from ringing in the reverb tail. Mode Enhancement works by continuously modulating certain delay lines (taps) within the program algorithms, which increases the effective density without thickening the reverb itself. Mode Enhancement is factory-optimized for each program and should not require adjustment in typical use.
The Pitch Shift controls are accessed in the Hidden Controls panel. See page 200 for access details. Decay Optimization Decay Optimization improves the Lexicon 224 reverb clarity and naturalness by dynamically reducing reverb diffusion and coloration in response to input signal levels. However, if set too high, it can make the decay less even. Decay Optimization has two control elements: Enable and Amount.
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. The default state is ON. Note: Solo is a global (per Lexicon 224 plug-in instance) control.
Access The hidden controls are exposed by clicking the “OPEN” text to the right of the Display 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 retained when a new Lexicon 224 plug-in is instantiated. Pitch Shift Pitch Shift is a component of Mode Enhancement. See “Mode Enhance Pitch Shift” on page 198 for parameter details.
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 overwriting the right channel value). When Link is active, automation data is written and read for the left channel only. The automation for the left channel controls both channels in Link mode.
Program Descriptions P1 Small Concert Hall B This program emulates the sound of a small concert hall, with moderate initial 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 recommended instead). The most natural sound is obtained when Bass and Mid are relatively close to the same setting. This program uses the exact same algorithm as P3 Large Concert Hall B.
P8 Constant Density Plate A In naturally occurring reverb, new reflections are continuously added to the decaying sound over time. This sonic build-up increases density and coloration in the reverb tail. P8 Constant Density Plate A has high initial density and coloration (giving a “plate” type of sound), however the density does not increase over time and remains inherently constant. This can result in less “swoosh” in the reverb tail and provides another creative option.
Table 17. Lexicon 224 Outputs Used With Monophonic Output Program Output(s) 1. Small Concert Hall B 2. Vocal Plate A A+C 3. Large Concert Hall B A 4. Acoustic Chamber A+C 5. Percussion Plate A A+C Default Parameter Values Program Output(s) 6. Small Concert Hall A A 7. Room A A 8. Constant Density Plate A A 9. Chorus A A+C Table 18 below lists the default values of all available parameters for each program. Table 18.
CHAPTER 19 Little Labs IBP Overview The Little Labs IBP Phase Alignment Tool easily eliminates the undesirable hollow 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” tool, but as a device for manipulating audio phase as a creative, tonal color tool as well.
Little Labs IBP Controls All parameters are clearly labeled with control names. Please refer to Figure 64 on page 206 for control descriptions. Delay Adjust The Delay Adjust parameter is unique to Universal Audio’s “workstation” version 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 90°/180° This switch determines the range of the Phase Adjust parameter. This is useful 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°. Phase Center Lo/Hi This switch sets the range of frequency emphasis. When the switch is disengaged (lighter), the Phase Center range is Hi. When the switch is engaged (darker), the Phase Center range is Lo.
CHAPTER 20 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 Little 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 65. 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. In Use The VOG is intended for mixing, mastering, post-production sweetening, sound design, and audio restoration. Use it to easily simulate proximity effect for adding chest resonance and “heft” to vocals.
Like the original hardware, increasing the Frequency value (rotating clockwise) 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 arbitrary and do not reflect a particular frequency value. Center The two Center switches define the active center frequency of the effect, which in turn determines the available frequency range.
The Little Labs VOG hardware unit UAD Powered Plug-Ins Manual - 213 - Chapter 20: Little Labs VOG
CHAPTER 21 Manley Massive Passive EQ Overview Universal Audio’s UAD Powered Plug-In versions of the Manley Massive Passive 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 — delivering a fundamentally different sounding EQ that is beyond compare.
Massive Passive Screenshots Figure 66. The Massive Passive plug-in window Figure 67. 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. Keeping these points in mind may help you obtain more satisfactory results. See “Notes from Manley Laboratories” on page 224 for more tips.
Unique Shelves Most EQs offer a shelving mode for the edge bands only. Massive Passive offers the shelving option on all bands for expanded sonic possibilities, such as “staircase” EQ curves. No negative feedback loops One result of not using negative feedback loops in the design is that the gain control for a band cannot have a “bipolar” boost and cut control. Only band gain is available; how that band gain is applied, either as a boost or as a cut, is specified with a separate toggle switch.
Standard vs. Mastering Versions The layout and function of the Massive Passive controls are essentially identical for both the Standard and Mastering versions. The exact control differences between the controls are detailed in Table 20 below. Table 20. Control differences between Massive Passive versions Standard Mastering Channel Gain Range -6 dB to +4 dB ±2.5 dB (0.
Shelf/Bell The Shelf/Bell toggle switch defines the shape of the filter band. A 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 219) affects the slope of the band filters in both Shelf and Bell modes.
Gain has a fair amount of interaction with the Bandwidth control. The maximum band gain is available in Shelf mode when Bandwidth is fully counter-clockwise; less band gain is available in Shelf mode as the Bandwidth is decreased (rotated clockwise). Conversely, the maximum gain is available in Bell mode when Bandwidth is fully clockwise; in Bell mode less band gain is available as Bandwidth is decreased (rotated counter-clockwise).
As Bandwidth is increased in Shelf mode, a bell curve begins to be introduced in the opposite direction (i.e., overshoot). For example, if the Shelf is boosted, a dip is created at higher Bandwidth values. At maximum Bandwidth, this overshoot curve is pronounced. The effect of the Bandwidth control in Shelf mode is shown in Figure 69 below. Figure 68. Effect of Bandwidth control on response curve in Bell mode Figure 69.
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 21 below. Frequency Available Frequencies Table 21.
The Channel Gain controls are intended to help match levels between “Bypass” 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. The range is small to allow easier and finer adjustments. Filters Low Pass and High Pass filters are available for both channels. The response curves of the filters are shown in Figure 70 below.
Figure 70. High Pass and Low Pass filter response curves (standard version) Mastering Filters The Low Pass/High Pass filter frequencies in the mastering version are tuned 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. Other Controls The Power and Link controls are global to both channels. Power is a two-state knob that determines whether the plug-in is active.
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 position (channel 1 & 2 controls are ganged together in Link mode). When Link is active, automation data is written and read for channel one only. In this case, the automation data for channel one will control both channels.
• 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. On the other hand, because of how transparent this EQ is, you might actually be EQing more than you could with a different unit.
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 recommended reading for those interested in technical details. The manual can be found on their website, along with info about their other great products: • http://www.manley.com/manuals.
CHAPTER 22 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. The Moog Multimode Filter is a ‘digital-only’ tabletop filter set that combines the best of Bob Moog’s classic designs with select features from his final Voyager instrument.
Moog Filter Screenshot Figure 71. The UAD Moog Filter plug-in window Moog Filter Controls The Moog Filter is true stereo, with separate filters for the left and right channels. 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. Drive Drive controls the amount of saturation gain before the filter. Drive is where much of the sonic “juice” in the UAD Moog Filter originates.
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 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. Drive/Gain LED Envelope The Envelope controls (Envelope knob, Smooth/Fast switch) closely mimic the controls of the MF-101 Moogerfooger.
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 available range of 12 Hz to 12 kHz on the Moog Mulitmode Filter. In lowpass mode, frequencies above the cutoff are attenuated. In highpass mode, frequencies below the cutoff are attenuated.
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 cutoff 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. Note: When set to Track, the plug-in does not “track” the input signal frequency like a synthesizer filter.
LFO The LFO (low frequency oscillator) modulates the filter cutoff frequency. Several waveform shapes are available. The LFO can be synchronized to the tempo of the host (see Free/Sync below). Amount Amount controls the depth of the LFO filter cutoff modulation. A higher value will have a broader filter sweep. Rate Rate controls the speed of the LFO. The available range is from 0.03 Hz to 25 Hz in Free mode, or 16/1 to 1/64 to in Sync mode.
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. Mix Mix varies the amount of filtering that is occurring. It is not a true dry/wet control; it mimics the mix function on the MF-101 Moogerfooger.
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 provide sonic characteristics very similar to the Moog Filter but with significantly less DSP usage. It is provided to allow Moog Filter benefits when DSP resources are limited. The UAD Moog Filter SE sounds great even without Drive, and is very usable in many situations.
Moog Filter Latency The Moog Filter (but not the Moog Filter SE) uses an internal upsampling technique 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. The Moog Filter SE does not require additional latency compensation because it is not upsampled.
CHAPTER 23 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 73. 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 oscillator (LFO). The delayed signal can be mixed with itself in a feedback loop (“regenerated”), and its polarity can be inverted.
Software-Only Features The UAD MXR Flanger/Doubler plug-in has some features not included in the 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; and stereo output can be summed to mono. Stereo Functionality The original hardware is monophonic. To accommodate modern applications, the plug-in can be used in mono-in/stereo-out and stereo-in/stereo-out configurations.
Flanger When in the “down” (gray) position, Flanger mode is active. This is the default setting. Doubler Doubler mode is active when the button is in the “up” (white) position. Stereo Mode This software-only switch modifies the processed signals at the outputs when used in a stereo-output configuration. The control does not switch the processor between mono and stereo modes; both modes are true stereo (when configured for stereo output).
This continuous control determines the delay time of the processor. The delay time is modulated by the Sweep LFO when the Width value is higher than 0%. Manual The available range of the control depends on the setting of the Effect button. In Flanger mode, the available delay time range is 4.9 milliseconds to 0.33 milliseconds. In Doubler mode, the available delay time range is 66 milliseconds to 18.5 milliseconds.
This continuous control adjusts the blend between the original dry signal and the processed wet signal(s). The available range is 0 – 100%. Mix 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). When Mix is set to the minimum/dry position, the input signal is colored by the electronics of the unit (like the original hardware).
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. Sync See Chapter 8 “Tempo Sync” in the UAD System Manual for complete details about this feature. Rate Display The rate of the Sweep LFO is displayed here. When Sync is inactive, the LFO speed is displayed in Hertz.
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 application supports full plug-in delay compensation throughout the signal path, or when it is used only on the outputs.
CHAPTER 24 Neve 1073 Equalizer Overview Designed by the Rupert Neve company in 1970, perhaps no other studio tool is as ubiquitous or desirable as the Neve 1073 channel module. Without exaggeration, Neve consoles such as the 8014 (where the 1073 originated) have been used on a majority of popular recordings of the late 20th century, and the 1073 easily tops the short-list of audio design masterpieces.
The Input Gain control sets the level at the input of the plug-in. The range is from –20 dB to +10 dB. Input Gain 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). Clicking the OFF screen label toggles between OFF and the previously set Input Gain value. You can also click the Neve logo to toggle between OFF and the previous state.
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 disabled. UAD CPU usage is not reduced when the band is OFF. The low frequency band is controlled by dual-concentric knobs, delivering smooth shelving equalization. Low Band The inner knob controls the band gain, and the outer ring selects the frequency or band disable. These two controls are detailed below.
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. EQL 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) position. Click the button to toggle the state. In the hardware 1073, the audio is still slightly colored even when the EQL switch is in the Out 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” in the UAD System Manual for more information. The Neve 1073SE does not require additional latency compensation because it is not upsampled.
CHAPTER 25 Neve 1081 Equalizer Overview The Neve 1081 channel module was first produced in 1972 by Neve, and was used to provide the mic/line amp and EQ sections in consoles such as the Neve 8048. Vintage 8048 consoles, with 1081 modules, are still in wide use today at classic facilities such as The Village in Los Angeles, and have been chosen by artists ranging from The Rolling Stones to The Red Hot Chili Peppers.
Neve 1081 and 1081SE Controls Overview The Neve 1081 channel module is a four-band EQ with high and low cut filters. The 1081 features two parametric midrange bands, with “Hi-Q” selections for tighter boosts or cuts. Both the high and low shelf filters have selectable frequencies and may be switched to bell filters. Other features include a –20 to +10 dB input gain control, phase reverse, and EQ bypass. The bands are arranged and grouped as in Figure 77 below. The bands feature dual-concentric controls.
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 Band High Gain The equalization gain for the high band is selected with the inner knob of the dual-concentric control. Rotate the control clockwise to add the famous high-end Neve sheen, or counter-clockwise to reduce the treble response. The available range is approximately ±18 dB.
High-Mid Frequency The high-midrange band frequency is selected with the outer ring of the dualconcentric 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. You can also click the midrange symbol below the knob to cycle through the available values, or shift + click to step back one frequency. Note: The available high-mid band center frequencies are 1.5 kHz, 1.8 kHz, 2.2 kHz, 2.7 kHz, 3.3 kHz, 3.9 kHz, 4.
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 reduced when the band is OFF. Low-Mid Q Select The High Q button switches the response of the low-mid band from “normal” to a narrower bandwidth for a sharper EQ curve. The band is in normal mode by default; it’s in high Q mode when the button is “down” (darker).
The independent low and high cut filters are controlled by the dual-concentric knobs to the right of the low band (see Figure 77 on page 250). The controls specify the fixed frequency of the cut filter. The cut filters have an 18 dB per octave slope. Cut Filters Click+drag the control to change the value, or click the “silkscreen” frequency values. Note: You can also click the high cut/low cut symbols below the knob to cycle through the available values, or shift + click to step back one frequency.
Neve 1081SE Figure 78. 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. Nobody with “golden ears” will say it sounds exactly like the 1081, but it still sounds great and is very usable in most situations.
CHAPTER 26 Neve 31102 Console EQ Overview The Neve 8068 console, featuring the 31102 EQ, was used to hand-mix one of the best selling debut albums of all time; Appetite For Destruction by GunsN-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.
Neve 31102 and 31102SE Controls The Input Gain control sets the level at the input of the plugin, and doubles as a plug-in bypass control. The range is from –20 dB to +10 dB, and off. Input Gain When the Input Gain knob “snaps” to the off position, plug-in processing is disabled and UAD DSP usage is reduced. Note: (UAD-2 only) UAD-2 DSP usage is reduced only when DSP LoadLock is disabled. If DSP LoadLock is enabled (the default setting), setting Input Gain to off will not reduce DSP usage.
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 Band Midrange Gain The equalization gain for the midrange band is selected with the inner knob of the dual-concentric control. The available range is approximately ±15 dB. The gain value is zero when the knob position indicator is pointing straight down.
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 available low band center frequencies are 35 Hz, 60 Hz, 110 Hz, 220 Hz, and off.
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) position. Click the button to toggle the state. EQL In the hardware 31102, the audio is still slightly colored even when the EQL switch is in the Out position. This is due to the fact that the signal is still passing through its circuitry. Therefore, the signal will be slightly colored when this switch is in the Out position.
Neve 31102 Latency The Neve 31102 (but not the 31102SE) 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 other UAD Neve EQ’s. See Chapter 9 “UAD Delay Compensation” in the UAD System Manual for more information. The Neve 31102SE does not require additional latency compensation because it is not upsampled.
CHAPTER 27 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 81. 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 267). When the 33609 is used in a mono-in/mono-out configuration, the channel 2 controls are disabled. Each channel consists of a compressor and a limiter. Each of these functions has its own separate group of controls.
Modeling The UAD Neve 33609 models all aspects of the original 33609 hardware, including the desirable harmonic distortion characteristics. These qualities are more prominent at higher input levels (see “Headroom” on page 269” for more info). When the compressor and limiter are both disabled, some (good) coloration of the signal occurs, just like the hardware. If a true bypass is desired, use the power switch (“Power” on page 271) to disable the plug-in.
The automatic settings (a1 and a2) are program dependant. The value for a1 can be as fast as 40ms, but after a sustained period of high signal level, the period is ≈1500ms. The value for a2 can be as fast as 150ms, but after a sustained period of high signal level, the period is ≈3000ms. Limiter In This toggle switch enables the limiter portion of the plug-in. The limiter has no effect unless this switch is in the “In” (down) position.
Compressor Gain This makeup gain control increases the signal level out of the compressor to compensate for reduced levels as a result of compression. The available range is 0 to +20 dB, in 2 dB increments. Make sure to adjust the Gain control after the desired amount of compression is achieved with the Threshold control. The Gain control does not affect the amount of compression. Note: If the limiter is also enabled, this gain is applied before the limiter stage.
Note: The meter indicator moves farther to the right as more gain reduction is occurring. This meter behavior is opposite that of many compressors. Link This switch is a software-only addition that allows the two sets of controls for each channel to be linked for ease of operation when both channels require the same values, or unlinked when dual-mono operation is desired. The Link parameter is stored within presets and can be accessed via automation.
Stereo In stereo mode, the left channel is fed to the channel one compressor, and the right channel is fed to the channel two compressor. The two compressors are constrained so that they both compress the same amount at any instant. This prevents transients which appear only on one channel from shifting the image of the output. Any big transient on either channel will cause both channels to compress.
Headroom Background The hardware Neve 33609 can accept an analog signal level of approximately +26 dBu before undesirable signal clipping occurs. As the signal increases up to this point however, desirable audio-path nonlinearities and “good” harmonic distortion characteristics occur. This musically pleasing “warmth” at higher levels is what gives the unit much of its revered sonic character.
22 dB Typical starting point for individual track inserts where maximum gain reduction is desired. This setting is equivalent to having a real hardware 33609 connected to a +4 interface with +22 dB headroom. 18 dB Typical starting point when used on a buss or group where nominal gain reduction is desired. This setting is equivalent to having a real hardware 33609 connected to a +4 interface with +18 dB headroom. 14 dB Typical starting point for mastering where minimal gain reduction is desired.
Power 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 plugin 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; the switch is illuminated in red when the plug-in is active. You can click-hold the power switch then drag it like a slider to quickly compare the enabled/disabled state. Note: Neve 33609SE Figure 83.
Compensating for Neve 33609 is not required if the host application supports full plug-in delay compensation throughout the signal path, or when it is used only on the outputs. Note: All visual and aural references to the Neve® 1073, 1081, 31102, 88RS, and 33609 products and all use of AMS-NEVE’s trademarks are being made with written permission from AMS-Neve Limited.
CHAPTER 28 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 Screenshot Figure 84.
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. Signal Flow The output of the cut filters is fed to the input of the dynamics or EQ section (dependent upon the Pre-Dyn switch).
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 86 on page 276), and hysteresis is available (see Figure 87 on page 277). Expansion mode is enabled by rotating the hysteresis (HYST) control fully counter-clockwise (or clicking the EXP label).
The Hysteresis knob sets the difference in threshold for signals that are either rising or falling in level. Signals that are rising 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. Up to 25 dB of hysteresis is available. See Figure 87 on page 277.
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 278). 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 percussion and other instruments with very fast decays. Using fast settings on other sources may produce undesirable results. Note: This meter displays the amount of gain attenuation (downward expansion) occurring in the gate/expander module.
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. The compressor will still release somewhat during the snare hits, however. For additional information, see “The LA-2 captures one of the earliest Teletronix examples.
Threshold defines the input level at which limiting or compression begins. Signals that exceed this level are processed. Signals below the threshold are unaffected. L/C Threshold The available range is +20 dB to –10 dB. A range of 0 dB to –30 dB is available when the –20 dB switch is engaged (see “L/C Threshold –20 dB” on page 281). As the Threshold control is increased and more processing occurs, output level is typically reduced.
Release sets the amount of time it takes for processing to cease once the input signal drops below the threshold level. The available range is 10 milliseconds to 3 seconds, and automatic. L/C Release Automatic triple time-constant program dependent release time is activated by turning the release control fully clockwise (to 3s) or by clicking the “AUTO” label text.
With the UAD Neve 88RS EQ, the Q value and range is dependent on the gain setting of the band. With any non-zero gain setting, the Q will be calcuEQ gain module lated in real-time for that band. But if the band is zero, Q will always disenable switch play zero. “The unique sound of AMS-Neve EQ is the result of years of research and extensive studio experience.
You can use this button to compare the equalized signal to the original signal or bypass the EQ altogether. UAD DSP load is reduced when this module is inactive (unless UAD-2 DSP LoadLock is enabled). 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. HF Gain This control determines the amount by which the frequency setting for the HF band is boosted or attenuated. The available range is ±20 dB.
The Q (bandwidth) control defines the proportion of frequencies surrounding the HMF band center frequency to be affected by the band gain control. The filter slopes get steeper (narrower) as the control is rotated clockwise. The available range is 0.4 to 10. HMF Q LMF Freq This control determines the LMF band center frequency to be boosted or attenuated by the LMF Gain setting. The available range is 120 Hz to 2 kHz.
Cut Filters In addition to the four-band EQ, UAD Neve 88RS offers two cut filters, one each for low and high frequencies. The slope of the cut filters is 12 dB per octave. Each cut filter has two controls: Cut Enable and Frequency. Both controls are detailed below. Note: UAD DSP load is not reduced when the cut filters are disabled. This button activates the cut filter. The cut filter is active when the button is gray and the red indicator illuminates.
The Phase (Ø) button inverts the polarity of the signal. The signal is inverted when the button is gray and the red indicator illuminates. Leave the button inactive (unlit) for normal phase. Phase Output The Output knob controls the signal level that is output from the plug-in. The default value is 0 dB. The available range is ±20 dB. Power The Power switch determines whether the plug-in is active.
CHAPTER 29 Ocean Way Studios Dynamic Room Modeling Welcome to Ocean Way Studios — the World’s First Dynamic Room Modeling Plug-In Developed by Universal Audio and Allen Sides, the Ocean Way Studios plug-in re-writes the book on what’s possible with acoustic space emulation.
dispersion patterns. This enables users to recreate the same set-ups used to record some of the biggest acts of all time — including Michael Jackson, Madonna, U2, Ray Charles, Radiohead, Beck, Tom Petty, The Rolling Stones, and many, many more. Even the unique consoles in each room have been incorporated into the overall sound — the Ocean Way Recording-modified Focusrite ISA 110 console in Studio A, and the famous custom-modified Putnam/Dalcon console in Studio B. Ocean Way Studios Screenshot Figure 89.
What Is Ocean Way Studios? Ocean Way Studios is a dynamically adjustable room emulator for adding the ambience of Ocean Way Recording’s acclaimed studios A or B to audio signals. Figure 90. Interior photos of Room A (left) and Room B (right) at Ocean Way Recording Ocean Way Studios offers two modes of operation.
Hybrid Technology Ocean Way Studios is not a general impulse response (“IR”) convolution reverb nor a typical algorithmic reverb. Instead, Ocean Way Studios utilizes breakthrough hybrid technologies, combining expertly sampled impulse responses with advanced algorithmic DSP techniques.
Operational Overviews Overviews of important underlying concepts are presented below. For details about how to operate the specific controls, see “Ocean Way Studios Controls” beginning on page 304. Modes Overview Ocean Way Studios offers two modes of operation: Re-Mic and Reverb. These modes process signals in fundamentally different ways.
It’s important to notice that the recorded direct signal component (#1 on previous page) is different than pre-existing dry (unprocessed) acoustic recordings in a DAW. This is because within a DAW, the dry audio was already recorded – so it already contains the direct signal component (along with all the other components) that was captured by the microphone originally used. This distinction is fundamental to the Re-Mic process.
Microphones Overview In addition to the studio room acoustics, the microphones used in the development of Ocean Way Studios are a significant contributor to the tonality and fidelity of the plug-in. Microphone Selections Ocean Way Studios contains 11 different microphone pairs. Additionally, some of these microphone pairs are available with cardioid and omnidirectional polar frequency response patterns. The microphones that are available, along with their descriptions, are listed in Table 22 below.
Polar Patterns All microphone selections are denoted with O, C, or 8 after their name. Microphones with “O” after the name indicates the polar frequency response pattern of the mic is omnidirectional. Microphones with “C” after the name indicates the polar pattern of the mic is cardioid. Microphones with “8” after the name indicates the mic has a “figure 8” polar pattern.
Distance The distance from the microphone pair to the source can be dynamically adjusted using the Distance control. Just as when recording with microphones in the physical realm, the mic-to-source distance can have a significant impact on the sound that is captured. The room will sound tighter and more present the closer the mics are to the source; conversely, the room gets “bigger” when the mics are farther away from the source.
Removing this inherent delay can be especially useful in these scenarios: • If a source is recorded with a distant room mic, it will play back later in relation to sources that are close-miced. Typically, this delay can be compensated in the DAW by manually shifting the track forward in time so it aligns with the other instruments. With Ocean Way Studios, removing the microphone delay will align the distant mics automatically.
Using Ocean Way Studios For details about how to operate the specific controls, see “Ocean Way Studios Controls” beginning on page 304. Best On Dry Sources Ocean Way Studios does not remove already-recorded ambience from existing audio signals. For optimum ambience control when using the plug-in, the source audio should be as dry as possible. However, Ocean Way Studios is very forgiving, and great results can be obtained even if the original source has existing ambience.
When to use Re-Mic Mode Use Re-Mic mode to “replace” existing audio with new audio that inherits the sonic characteristics of Ocean Way Studios. The original dry signal component is removed and completely immersed with Ocean Way room sound. See “Re-Mic Processing” on page 293 for an overview of Re-Mic mode. Figure 94 shows how to configure the Re-mic workflow in a DAW. In this example, all the drums are routed to a submix bus instead of the main outputs.
Dual-Mode Example Figure 95 shows how to use both Re-Mic and Reverb modes with two instances of the plug-in, combining the workflows of the two previous examples. The illustration combines a drum submix is being used for Re-Mic mode, while send/return routing is being used for guitar and vocals in Reverb mode. Figure 95. DAW signal routing in a workflow with two Ocean Way Studios plug-in instances. One uses Re-Mic mode for the drum submix, and the other uses Reverb mode for the guitar and vocals.
Whenever more than one microphone pair is simultaneously enabled, careful attention should be paid to the Distance (position) and Polarity Invert parameters to avoid potential phase issues. Just as with moving microphones around and changing signal polarity when recording acoustically, changes to Distance and Polarity Invert can have a dramatic effect on the sound. Note that sometimes phasing can sound just fine, and can be useful for creative purposes. Ocean Way Studios sounds amazing when set properly.
The latency of Ocean Way Studios depends on the sample rate. The exact latency values are provided in Table 23. Table 23. Latency in Ocean Way Studios Sample Rate (kHz) Latency (samples) Latency (time) Sample Rate (kHz) Latency (samples) Latency (time) 44.1 192 4.3 ms 96 688 7.1 ms 48 192 4.0 ms 176.4 1568 8.9 ms 88.1 688 7.8 ms 192 1568 8.2 ms Note: As with all UAD plug-ins, the latency of Ocean Way Studios is automatically compensated by the DAW.
Table 24.
Ocean Way Studios Controls Mode Ocean Way Studios offers two modes of operation: Re-Mic and Reverb. Click a mode control to activate the mode. The button of the current mode is illuminated. For details about the differences between these two modes, see “Modes Overview” on page 292. Re-Mic In Re-Mic mode, the dry signal path is eliminated and the audio is processed as if it was recorded inside Ocean Way Recording.
OWR A Ocean Way Recording A is Ocean Way's most spacious studio (45’ x 52’), suitable for four-piece bands to full orchestras. With a rich sound, exceptionally clear low end, and super smooth decay, OWR A is a study in classic studio design. Associated artists include John Mayer and Whitney Houston, to classics like Frank Sinatra and Count Basie. OWR B Ocean Way Recording B is M.T. “Bill” Putnam’s crowning achievement in studio design.
Source A variety of audio sources (dispersion patterns) were modeled for Ocean Way Studios. The Source menu sets the optimum placement of the source within the room, as determined by the expertise of Allen Sides. Because an audio source’s placement within a room determines the dispersion pattern of sound waves throughout the room, the active source can have a significant impact on the sound in the room.
Display Panels The Display Panels show helpful information about the current state of the plug-in. The four available panels are shown in Figure 97. Click the buttons beneath the Display Panels to choose one. The button of the currently active panel is illuminated. Note: The Display Panels are for informational purposes only. There are no parameter controls within any of the Display Panels, and the Panel selection controls cannot be automated. Figure 97.
The locations of the source and microphones within the room are determined by the Source and Distance parameters. Note: Microphones that are muted are not shown in the Position Display Panel. Master EQ The Master EQ panel displays the state of the Master EQ settings. When the Master EQ is disabled (or when both Master EQ Gain values are zero), the frequency spectrum is flat. Interior The Interior panel displays a photograph of the currently selected studio.
Mic Selection The microphones used in the room are selected with this menu. To change the active microphone, click the current microphone name then select the desired mic from the drop menu, or click the microphone image to cycle through the available microphones. Tip: To maintain the current Distance value for the selection when changing microphones, press Shift on the computer keyboard when making the mic selection. Not all microphones are available for all sources.
Cut Filters Independent High Cut and Low Cut filters can be enabled on each microphone. Click the switch to toggle the filter state. The filter is active when the switch is illuminated. The cutoff frequency and filter slope varies for each of the microphones, as shown in Table 25 below. Table 25.
Gain This fader controls the volume level of the microphone. Gain has a logarithmic taper for a more musical response. The gain range is off to +12 dB. Gain is at unity when set to the zero position. Tip: To quickly return to the 0 dB (unity) position, click the associated NEAR/MID/FAR text label beneath the fader, or the associated “0” text label at the fader’s unity gain position.
Mono Output When the plug-in has monophonic output (when in a mono-out configuration or set to mono with the Mono switch), the microphone icon(s) in the Position panel shows a single icon for each active microphone, as shown at right. This is a convenient visual reminder that the plug-in output is monophonic. Stereo Output When the plug-in has stereo output, the microphone icon(s) in the Position panel shows dual icons representing the matched stereo set for each active microphone pair, as shown at right.
Wet Solo defaults to On, which is optimal when using Ocean Way Studios in Reverb mode in the “traditional” reverb configuration (placed on an effect group/bus that is configured for use with channel sends). When Ocean Way Studios is used on a channel insert in Reverb mode, this control should be deactivated so the Dry/Wet mix can be adjusted. Wet Solo is fixed in the enabled position in Re-Mic mode so the original dry signal cannot be mixed with the modeled direct signal component within the plug-in.
Master EQ This group of parameters contains the controls for Ocean Way Studio’s master equalizer. It is a two band (low and high) shelving EQ that uses analog-sounding algorithms for great tonal shaping options. The slope of both filters is 12 dB per octave. The Master EQ section is independent from the reverb algorithms. A graph of the current curve is displayed in the Master EQ display panel.
The History of Ocean Way Recording Ocean Way Recording in Hollywood California is the world's most awarded studio complex. Albums recorded at the studio have sold over 1 billion units. Generations of music icons, from Frank Sinatra, Nat King Cole, Ray Charles to The Rolling Stones, Eric Clapton, and Michael Jackson, all the way to contemporary artists like Green Day, Dr.
Putnam moves to Los Angeles In order to be a proper studio, Sides needed a recording console. This is where the story of Ocean Way truly begins; How Sides ended up purchasing Western Recorders' original tube console and came face to face with M.T. “Bill” Putnam. Putnam was a true renaissance man in the world of sound and music. His combined skills as a record producer, audio engineer, songwriter, singer, electrical engineer, inventor, studio owner and businessman are unparalleled to this day.
“One man's junk is another man's treasure; and in this case, I was able to acquire some old Fairchild limiters, UA tube limiters, Macintosh tube amps, and enough equipment to completely fill my garage studio. It was the deal that really put me in business. However, there was a slight problem. I didn't actually have the 6 grand, so I wrote a check, picked up the stuff, and within six hours had sold enough gear to cover my check.
of Allen's best long term clients. A couple years later, Bill sold United/Western to Allen, at which time United Recorders then became Ocean Way Recording. It was also during this time Sides began buying close to a thousand tube microphones from overseas: The European studios and broadcasters were dumping loads of “antiquated” tube mics for brand new phantom-powered transistor mics. He carefully went through every mic, picking the absolutely best of the best and selling off the rest.
UAD Powered Plug-Ins Manual - 319 - Chapter 29: Ocean Way Studios
CHAPTER 30 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.
Precision Buss Compressor Screenshot Figure 98. 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. Filter regulates the cutoff frequency of the filter on the compressor's control signal sidechain.
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. When Ratio is changed, Threshold numerical values are updated but the Threshold knob position does not move. Note: As the Threshold control is decreased and more compression occurs, output level is typically reduced.
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. Fade The Precision Buss Compressor provides a Fade function that, upon activation, automatically reduces the plug-in output to minimum within a specified time period.
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. Note: Shift+click the Fade button to instantly return the level back to 0 dB (this feature cannot be automated).
The Gain Reduction meter displays the amount of gain reduction occurring within the compressor. Gain Reduction Meter 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. The Power switch determines whether the plug-in is active. Click the toggle button or the UA logo to change the state.
CHAPTER 31 Precision De-Esser Overview The Precision De-Esser seamlessly and accurately removes sibilance from individual 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 twoposition “Speed” button gives control over the envelope (attack and release) of the detector.
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 the signal level at which the processor is activated. Rotate Threshold counter-clockwise for more de-essing.
Note: When Solo is active, changes to the Threshold and Split controls cannot be heard. Width controls the bandwidth of the de-essing sidechain when in bandpass mode. Bandpass mode is active when the control is in any position except fully clockwise. Width Smaller values have a narrower bandwidth, causing a tighter, more focused de-essing effect. Higher values have wider bandwidth, for de-essing when undesirable frequency ranges are broader.
The Gain Reduction meter provides a visual indication of how much attenuation (compression) is occuring. Signal peaks are held for 3 seconds before resetting. Gain Reduction When Split is on, the amount of sidechain attenuation is displayed. When Split is off, it displays the attenuation of the entire signal. The Power switch determines whether the plug-in is active.
CHAPTER 32 Precision Enhancer Hz Overview The Precision Enhancer Hz allows the user to selectively add upper harmonics to bass fundamentals, sometimes referred to as “phantom bass.” This significantly enhances the perception of low-end energy beyond the conventional frequency response of small speakers. These harmonics stimulate a psychoacoustic bass-enhancing effect in the listener, giving even the smallest speakers greater translation of low frequency sources.
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 processing that occurs in the plug-in. The available range is from 0.00 to 100.0%. Effect Knob Technically speaking, Effect scales the input to the enhancer. Increasing this parameter makes the enhancer have a higher amplitude output for a given input level.
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. Mode C also works well on sub-mixes and program material. Moderate compression is applied to the harmonics signal, increasing the amplitude of the harmonics and altering their timbre.
Through filter isolation of the original bass content, the Hz Frequency parameter defines the cutoff frequency for the enhancement process. Frequencies below this value are enhanced by the processor. The available range is 16 Hz to 320 Hz. Hz Frequency Hz Solo Hz Solo isolates the original bass signal and can be combined with Effect Solo. Hz Solo is active when the button is red. Output controls the signal level that is output from the plug-in. The available range is –20 dB to 0 dB.
Precision Enhancer Hz Usage Notes • The Precision Enhancer Hz effect can serve multiple purposes. When the frequency 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. With the frequency control set to mid to higher frequencies, the effect is designed to add bass tone that would ordinarily disappear on smaller speakers.
CHAPTER 33 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 101. 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. The Effect Knob controls the amount of processing that occurs in the plug-in. The available range is from 0.00 to 100.0%. Effect Knob Technically speaking, Effect scales the input to the enhancer.
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 enhancement amount is increased as the input signal level increases. The kHz Frequency parameter is disabled in this mode. All Mode “All” mode is selected by shift+clicking Mode letters or LEDs. All Mode expands all frequencies of the input signal.
Generally speaking, adjust the Output control after the desired amount of processing 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. When the plug-in is disabled with the plug-in Power switch (but not the host plug-in enable switch), the output meters still function.
CHAPTER 34 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. The Precision Equalizer is based on industry standard analog mastering filters, and uses the classic parametric controls arrangement.
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) responses per band.
Dual Mode In Dual mode (dual-mono mode), the left and right parameters can be independently adjusted so that each side of the stereo signal can have different EQ settings. Note that this mode is infrequently used during mastering because phase, imaging, and level inconsistencies may be induced in the resulting stereo signal.
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). Power Switch Click the rocker switch to change the Power state. Alternately, you can click the blue UA logo to toggle the Power state. Band Controls Each control set (L and R) has four EQ bands.
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. Compensating for Precision Equalizer is not required if the host application supports full plug-in delay compensation throughout the signal path, or when it is used only on the outputs.
CHAPTER 35 Precision K-Stereo Ambience Recovery Psychoacoustic Ambience Recovery and Stereo Processor Created by Mastering Engineer Bob Katz Universal Audio’s Precision K-Stereo™ Ambience Recovery plug-in is a psychoacoustic processor conceived and created by famed mastering engineer Bob Katz.
Operational Overview Ambience Recovery The primary function of Precision K-Stereo is for ambience recovery and enhancement. 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. Presets Precision K-Stereo includes factory presets designed by Bob Katz.
Ambience Recovery Refer to Figure 104 below for the controls in this section. Figure 104. The Ambience Recovery controls Recover Enable This switch enables/disables the ambience recovery process. See “Ambience Recovery” on page 346 for an overview. Recover Enable must be engaged for the Ambience Gain, Enhance Deep/Wide, and Ambience Filters controls to have an effect. Note: This switch does not have any effect on the M/S Gain, L/R Gain, or Power controls.
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. Note: Controls in the Ambience Filters section have no effect unless the Recover Enable switch is engaged. Refer to Figure 105 below for the controls in this section. Figure 105.
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. The available range is 0.5 to 3. Smaller Q values cause the bell filter to effect a broader portion of the frequency spectrum, while high Q values affect a narrower spectrum.
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 ambience recovery or enhancement. Note: The Mid/Side controls are disabled when the plug-in is used in a mono-in/mono-out configuration.
Output Gain Controls Refer to Figure 107 below for the controls in this section. Figure 107. The Output Gain controls L/R Gain Enable This switch enables/disables the Left/Right Gain and Link controls. Link This switch links (gangs) the Left/Right Gain controls for ease of operation when both channels require the same value. Disable Link when independent left/right control is desired. Note: When Link is inactive and Link is engaged, the left gain value is copied to the right gain.
Output Level Meters The stereo peak/hold meters display the signal level at the output of the plugin. The meter range is from –30 dB to 0 dBFS. Signal peaks are held for 3 seconds 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). Click the switch or the OFF/ON labels to change the setting, or click the UA logo to toggle the setting.
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. Katz. Two categories are included: “MIX” presets for use during mixing, and “MSTR” for use during mastering.
BK-MSTR-Full Orchestra B Same as Full Orchestra A, but the high cut filter is set to 10 kHz which softens the high frequency ambience to tighten the percussion. BK-MSTR-Clear Presence If the hall or chamber in the original recording is missing some presence, here’s a suggestion on what to do. BK-MSTR-Raise Mid Instruments This preset subtly raises the Mid level 1 dB above the side level and recovers a bit of the spacious ambience which is commonly lost when the mid/side ratio is raised.
CHAPTER 36 Precision Limiter Overview The Universal Audio Precision Limiter™ is a single-band, look-ahead, brickwall limiter designed primarily for mastering with program material. The easyto-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 108. The Precision Limiter plug-in window Controls Overview Control knobs for the Precision Limiter behave the same way as all UAD plugins. 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. Isolated peaks will have a fast release time, while program material will have a slower release. Note: You can type “A” or “a” to enter Auto mode during text entry. Mode The Mode switch affects the attack shape of the limiter.
with 20-20 kHz pink noise on an SPL meter set to C-weighted slow (i.e. average) 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 monitor SPL.
Each of these modes displays the The RMS and instantaneous peak levels, which follow the signal, and the peak-hold level (see “Meter Response” on page 361). PK-RMS K-20 K-14 K-12 Figure 109. Precision Limiter Meter Types K-20 K-20 mode displays 0 dB at –20 dB below full scale. K-20 is intended for material with very wide dynamic range, such as symphonic music and mixing for film for theatre. K-14 K-14 mode displays 0 dB at –14 dB below full scale.
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. Meter Response The main stereo Input/Output meter actually displays three meters simultaneously: The RMS and instantaneous peak levels, which follow the signal, and the “peak-hold” (also known as global peak) level.
Figure 110. 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. The ranges and response for each meter type and scale is detailed below. PK-RMS In Normal mode, the meter range is –60 dB to 0 dB with a linear response of 0.
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 affects 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. Clear The meter Peak Clear switch clears the meter peak value display. It affects both the peak LED’s and the peak text display. Precision Limiter Latency The Precision Limiter has a 1.
CHAPTER 37 Precision Maximizer Overview The Precision Maximizer is a dynamic impact processor that uniquely enhances 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 fatiguing artifacts typically associated with traditional dynamic processors.
Precision Maximizer Screenshot Figure 111. 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 entry. The stereo peak Input Meter displays the signal level at the input of the processor, after the Input control. Input Meter 0 dB represents digital full scale (0 dBFS).
The Shape knob is the primary saturation control for the Maximizer effect. It contours the harmonic content and apparent dynamic range of the processor by changing the small-signal gain of the saturator. The available range is 0–100%. Shape At lower settings, apparent loudness is not as dramatic but harmonic processing still occurs, producing a richer sound with minimal reduction of dynamic range.
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 UAD-2 DSP LoadLock is enabled). Note: The Limit function provides a second stage of soft-saturation just before the output control for the plug-in. It prevents digital “overs” by protecting the plug-in output from exceeding 0 dBFS.
The Output knob controls the signal level that is output from the plug-in. The available range is –12 dB to 0 dB. Output 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. When Precision Maximizer is used for CD mastering and it is the last processor in the signal chain, the recommended Output value is –0.10 dB The stereo peak Output Meter displays the signal level at the output of the plug-in.
Operating Tips • As a starting point for general loudness enhancement, set Precision Maximizer 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 signal.
CHAPTER 38 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 flexibility 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. This helps to visually associate parameters to the band that they affect.
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 379). The Band Controls for each of the five bands are identical. Only one set of Band Controls is displayed at a time. The control set for any particular band is displayed by selecting the band (see “Band Select” on page 372). Band Select Selecting a band causes the controls for that band to be displayed in the Band Controls area.
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 select the LMF band. Band Parameters Because the Band Controls for each of the five bands are identical, they are only described once. All Button The ALL button provides a facility to link controls and copy parameter values to all bands when adjusting the current band. Each of the Band Controls has an ALL button.
Relative mode is not available for the Type parameter because the available Type values are discrete. Click and shift-click both activate Absolute mode for Type. Note: 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. Shift-click the ALL button to enter Absolute mode; the button background changes to red.
COMPRESS When a band is set to Compress, the dynamic range of the band will be reduced (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 increased (dependent upon the band threshold and input level). GATE When a band is set to Gate, the band behaves as a gate. A gate stops the signal from passing when the signal level drops below the specified threshold value.
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 Release sets the amount of time it takes for processing to cease once the input signal drops below the threshold level.
EQ Display In the EQ Display, the entire audio spectrum from 20 Hz to 20 kHz is displayed along the horizontal axis. Gain and attenuation of the five band frequencies (up to ±12 dB) are displayed along the vertical axis. Figure 113. Precision Multiband EQ Display Band Curves The Band Curves show the relative frequency and gain settings of the bands. The sides of the colored curves are a representation of each band’s frequency settings, and the top of each curve represents the band’s gain setting.
Adjusting Gain The gain of a band can be adjusted by click-dragging 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. Adjusting Gain and cF If the cursor is moved slightly lower than the above example, the gain and center frequency can be adjusted simultaneously, without adjusting the bandwidth.
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 377). The frequencies for each band can be modified by entering the values directly and by manipulating the colored band curves. Frequency Values All band frequency values are always displayed. Values can be input directly using text entry.
Dynamics Meters Realtime display of Precision Multiband dynamics processing is shown in the Dynamics Meters. This area also contains the band enable 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. Dynamics processing for each band is indicated by light blue “LED-style” metering. Zero dB is at the center of the meter, and the range is ±15 dB.
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 in the EQ Display is highlighted. Solo Display In addition to the Solo buttons, you can also control-click a band in the EQ Display to put any band (or bands) into Solo mode. Note: Global Controls Input Level Meter The stereo peak/hold Input Meter displays the signal level at the input of the plug-in.
Output Level Meter The stereo peak/hold Output Meter displays the signal level at the output of the plug-in. Signal peaks are held for 3 seconds before resetting. Output Level Knob The Output Level knob controls the signal level that is output from the plug-in. The default value is 0 dB. The available range is ±20 dB. EQ Display Switch The EQ Display mode can be static or dynamic. The EQ Display switch determines the active mode. Click the switch to toggle the mode.
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 39 Pultec Passive EQ Collection Introduction The Pultec Passive EQ Collection is the final word in fastidious circuit reproduction of Pulse Techniques’ revered passive EQs, considered to be the most popular outboard studio equalizers ever made. With the UAD-2’s increased DSP power and ten years of UA’s technological advancements since the original Pultec and Pultec-Pro plug-ins, UA has revisited the Pultec family.
Pultec Passive EQ Collection Screenshots Figure 114. The Pultec EQP-1A plug-in window Figure 115. The Pulteq MEQ-5 plug-in window Figure 116. The Pultec HLF-3C plug-in window Figure 117.
Pultec Plug-In Family The complete Pultec family is comprised of five individual plug-ins, as seen on the previous page. Each variation has its own unique sonic characteristics. Pultec Passive EQ Collection The Pultec Passive EQ Collection (introduced in UAD v7.1) provides access to three historical and highly coveted revisions in the Pultec product line.
Pultec Legacy The Pultec EQP-1A Legacy and Pultec-Pro Legacy plug-ins (Figure 117 on page 385) are the original versions of our Pultec emulations that run on both UAD-1 and UAD-2 devices. They still have a great sound and are very usable, especially when there are not enough DSP resources to use the secondgeneration versions in the newer Pultec Passive EQ Collection.
Artist Presets The Pultec Passive EQ Collection includes artist presets from prominent Pultec users. Some of the artist presets are in the internal factory bank and are accessed via the host application’s preset menu. Additional artist presets are copied to disk by the UAD installer so they can be used within Apollo’s Console application. The additional presets can be loaded using the Settings menu in the UAD Toolbar (see “Using UAD Powered Plug-Ins” in Chapter 7 of the UAD System Manual).
EQ Enable This is the EQ enable control. Like the original hardware, the signal is still colored even when this switch is in the out (down) position, because the signal is still passing through the I/O circuitry. If a true bypass is desired, use the Bypass/Gain knob. Bypass/Gain The function of this control differs between the newer EQP-1A and the Legacy version, as described below. Pultec EQP-1A This dual purpose knob is an output gain control, and the plugin bypass control.
LF Boost This knob determines the amount of low shelf gain to be applied to the frequency set by the CPS switch. LF Attenuation This knob (“ATTEN”) determines the amount of low shelf cut to be applied to the frequency set by the CPS switch. Background In the documentation supplied with hardware version of the EQP-1A, it is recommended that both Boost and Attenuation not be applied simultaneously because in theory, they would cancel each other out.
HF Boost This knob controls sets the amount of gain for the high frequency portion of the equalizer. High Attenuation Controls HF Attenuation Frequency This switch (“ATTEN SEL”) determines the frequency of the high frequency attenuator. Three frequencies are available: 5, 10, and 20 KCS. To cycle through the available values, click the “ATTEN SEL” text label, or shift+click the text label to cycle through available values in reverse. These shortcuts are unavailable with the Legacy version of the plug-in.
EQ Enable This is the EQ enable control. Like the original hardware, the signal is still colored even when this switch is in the out (down) position, because the signal is still passing through the I/O circuitry. If a true bypass is desired, use the Bypass/Gain knob (MEQ-5) or the Enable switch (Pultec-Pro Legacy; see page 389). Low Peak Controls LM Frequency This switch determines the frequency of the low-midrange portion of the equalizer.
High Peak Controls HM Frequency This switch determines the frequency of the high-midrange portion of the equalizer. Five frequencies are available: 1.5, 2, 3, 4, and 5 KCS. To cycle through the available values, click the “PEAK” text label, or shift+click the text label to cycle through available values in reverse. These shortcuts are unavailable with the Legacy version of the plug-in.
The HLF-3C interface is very simple and includes only the three controls shown in Figure 120. Figure 120. The Pultec HLF-3C controls Enable This toggle switch is the plug-in bypass control. When in the down position (bypassed), plug-in processing is disabled altogether. The plug-in is engaged with the switch is in the up position.
History In 1951, Pulse Techniques introduced the first passive Program Equalizer, the EQP-1. The passive EQ filter designs were originally licensed from Western Electric. Founders Ollie Summerland and Gene Shenk made up the Teaneck, New Jersey operation of Pultec. These two men comprised the engineering, marketing, sales and production staff for the entire history of the company, and made every item to order, all by hand.
CHAPTER 40 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. Room Shape and Material RealVerb Pro provides two graphic menus each with preset Room Shapes and Materials. You blend the shapes and material composition and adjust the room size according to the demands of your mix.
RealVerb Pro Background Pan Direct Path Source Input Wet/Dry Mix EQ Delay Early Reflections Gain & Mute Pans & Distance Gain Output LateField Reverb Delay Figure 121. RealVerb Pro signal flow Figure 121 illustrates the signal flow for RealVerb Pro. The input signal is equalized and applied to the early reflection generator and the late-field reverberation unit. The resulting direct path, early reflection, and late-field reverberation are then independently positioned in the soundfield.
The RealVerb Pro user interface is similarly organized (see Figure 122). 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 and sizes in the Shape panel; early reflection predelay and overall energy is specified at the top of the Timing panel. The Material panel is used to select relative late-field decay rates as a function of frequency.
To configure the room shape and size: 1. 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. 2. Select a room shape from the second (right) pop-up menu. The selected shape appears in the right side of the Shape circle. Adjust the room size with the bottom horizontal slider. 3. Blend the early reflection patterns of the two rooms by dragging the Blending bar.
Second material First material Blending bar First material selector popup menu Second material selector pop-up menu First material Thickness control Second material Thickness control Figure 124. RealVerb Pro Material panel Note: While materials are used to control decay rates as a function of frequency, the overall decay rate of the late-field reverberation is controlled from the Timing panel (see Figure 126 on page 405). To configure the room material and thickness: 1.
4. 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. To use only one room material, drag the Blending bar so the material is set to 100%.
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 lowfrequency to high-frequency decay times is 10:1. This means that the high frequencies will decay 10 times faster than the low frequencies. At 200% thickness, 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 equalizer can be configured as shelf or parametric EQs, as well as hybrids between the two. Amplitude control, third band Amplitude controls, first and second bands Band Edge control, second band Band Edge control, third band Figure 125.
3. 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 frequency for the high-shelf, drag to the left with the Band-edge control for the third band. 4. To attenuate the frequencies above the shelf frequency, drag the Amplitude controls for the first and second bands up or down.
Early Reflections display Amplitude control Predelay control Late-Field Reverberations display Amplitude Control Predelay control Decay Time control Diffusion control Figure 126. RealVerb Pro Timing panel To adjust the timing of the early reflections: 1. Drag the Amplitude control for the early reflections up or down (from –80 dB to 0 dB) to affect the energy of the reflections. The Amplitude value is indicated in the text field at the bottom of the Timing panel. 2.
3. 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 Decay Time is indicated in the text field at the bottom of the Timing panel. 4. To affect how quickly the late-field reverberations become more dense, adjust the Diffusion control at the right of Late Reflection display in the Timing panel. The higher the Diffusion value (near the top of the display), the more rapidly a dense reverb tail evolves.
To pan the direct (dry) signal: 1. 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: 1. Drag the left and right slider handles to adjust the stereo width. The length of the blue slider is adjusted.
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 128. RealVerb Pro Levels panel Morphing All RealVerb Pro controls vary continuously using proprietary technology to smoothly transition between selected values. This capability enables RealVerb Pro to morph among presets by transitioning between their parameter sets.
When in Morphing mode (Figure 130 on page 409), non user-adjustable con 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). On an insert effect, the Mix will change back and forth between the two mix values of each preset. Figure 130.
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” at the bottom of the presets menu. Edits to any and all presets in the list are maintained separately within each instance of a plug-in within a session.
CHAPTER 41 Boss CE-1 Chorus Ensemble Overview The Boss CE-1 Chorus Ensemble is another classic effect faithfully reproduced by our ace modeling engineers. The CE-1 is considered by many to the definitive chorus effect, renowned for its rich and unique timbres. Even for the mix engineer, stomp boxes can provide “secret weapon effects” not found any other way. In 1976, BOSS originated the chorus effect pedal, and nobody has come close to matching the CE-1’s captivating chorus sound since then.
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. Click to enable/disable the chorus or vibrato effect. The effect that will be heard is determined by the Vibrato/Chorus switch. Normal/Effect Switch The active state is black text. The inactive state has gray text.
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 modern era, enabling a true stereo input. Note: This switch has no affect in a mono-in/mono-out configuration.
These two knobs control rate and depth of the vibrato effect when CE-1 is in vibrato mode. Vibrato Controls Depth Knob The depth knob controls the intensity of the vibrato effect. Rate Knob The rate knob controls the rate of the vibrato LFO. The rate is indicated by the the Rate LED indicator. Note: When in chorus mode, the vibrato controls have no affect. Power Switch This switch determines whether the plug-in is active.
CHAPTER 42 Roland Dimension D Overview The Roland SDD-320 Dimension D is another classic effect faithfully reproduced by our ace modeling engineers. The Dimension D is a one of a kind studio gem that adheres to the principle of doing one thing, and doing it extremely well. Its one and only function: some of the best sounding stereo chorus ever made. However, the Dimension D is more than a chorus, it is really a unique sound enhancer for adding spatial effects to mono or stereo sources.
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. Dimension Mode The Dimension Mode determines the effect intensity. Four different modes are available. Mode 1 is the most subtle effect, and Mode 4 is maximum intensity. Multiple Buttons True to the original hardware, multiple Dimension Mode buttons can be engaged simultaneously for subtle sonic variations of the four main modes.
CHAPTER 43 Roland RE-201 Space Echo Overview In 1973, Roland created the Space Echo system that utilized multiple play heads to create warm, highly adjustable echo effects, which added wonderful tape character and chaos to performances and recordings. The Space 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.
Roland RE-201 Screenshot Figure 133. The Roland RE-201 plug-in window Roland RE-201 Interface The RE-201 interface is true to the original hardware, with a few customizations 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. We’ve also added a “Tape Age” switch to emulate new and older tape, a Wet Solo control for use as a bus/send effect, and an output volume control for utility.
The VU is essentially an input meter, therefore it doesn’t react when the Echo/Normal switch is switched from Echo to Normal. Note: The Peak lamp and VU meter measure signal just after the input volume control. However, like the original hardware, echo intensity (feedback) is applied just before the level detection circuit. For this reason, the Intensity control will affect the level readings.
The affect of each knob position is detailed in Table 32 on page 420. Table 32. RE-201 Mode Selector Positions Mode Knob Position Active Tape Heads 1 2 3 REPEAT (echo only) 1 2 3 REVERB + ECHO 4 • 5 6 7 • • • • Active Reverb 8 • • • • • • 9 • • • REVERB ONLY 10 11 • • • Reverb • • • • • • • • Bass This knob controls the low frequency response in the tape echo portion of the signal. It does not affect the dry signal or the reverb signal.
This knob controls the time interval of the echo effect. Rotating the control clockwise will decrease the delay time, and counterclockwise rotation will increase the delay time. Repeat Rate The available delay times are as follows: • Head 1: 69ms – 177ms • Head 2: 131ms – 337ms • Head 3: 189ms – 489ms The head times available with this control are dependent upon the “Mode Selector” on page 419.
Echo Volume has no affect when the Mode Selector is in the “Reverb Only” position. Note: This 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. Toggle the switch to change the Power state. Power Switch Toggling the power switch will also clear the tape echo. This can be useful if the RE-201 is self-oscillating and restarting the feedback loop is desired.
Splice Normally, the splice on the tape loop comes around at regular intervals. This interval varies, and is determined by the selected Repeat Rate. Depending on what Tape Quality is selected, the splice can be subtle or obvious, and can work as a catalyst for chaos especially when the RE-201 is in a state of self-oscillation. This switch resets the location of the tape “splice” when the switch is actuated.
CHAPTER 44 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 extremely simple to operate. The technology behind the processor isn't as important as how it sounds. However, for those who desire a deeper understanding of the process, a deeper explanation of the underlying technology is presented at the end of this chapter (see “Technology” on page 431). Attack enables amplification or attenuation of the attack of a signal by up to ±15 dB.
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. Overload The Overload “LED” illuminates when the signal level at the output of the plug-in reaches 0 dBFS. The indicator matches the behavior of the original hardware unit.
Acknowledgement 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 technology, SPL has graciously authorized us to use their documentation as well. The remainder of this chapter is excerpted from the SPL Transient Designer (RackPack) User Manual, and is used with kind permission from SPL. All copyrights are retained by SPL.
• 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 remarkably. Since muffling of any drum also changes the dynamic response, the Transient Designer opens up a whole new soundscape.
Guitars Use the Transient Designer on guitars to soften the sound by lowering the ATTACK. 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. Simply increase the ATTACK to get a clearer sound with more precision and better intonation despite any distortion. Heavy distortion also leads to very long sustain.
channels of the reverb return through two separate Transient Designer instances. 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, nothing shall be concealed from the more curious users. Differential Envelope Technology (DET) SPL’s DET is capable of level-independent envelope processing and thus makes any threshold settings unnecessary. Two envelopes are generated and then compared. From the difference of both envelopes the VCA control voltage is derived.
Figure 136 on page 432 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 amplitude of the attack is increased if positive ATTACK values are set. Negative ATTACK values reduce the level of the attack transient. Figure 136.
The SUSTAIN Control Circuitry The SUSTAIN control circuitry also plays host to two envelope generators. The 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. The control voltage of the VCA is again derived from the difference between the two voltages. Sustain amplitude is increased for positive SUSTAIN settings and reduced for negative settings.
Figure 140 on page 434 displays the processed waveforms with maximum and minimal sustain to compare against the original waveform in diagram 4. Figure 140. SPL Transient Designer Processed Sustain SPL Sound Performance Lab® and Transient Designer® are registered trademarks of SPL Electronics, GmbH Germany and are used under license. Portions of this SPL Transient Designer manual section is ©copyright SPL Electronics GmbH Germany and are used under license with kind permission from SPL.
CHAPTER 45 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 characteristics and intuitive EQ — rich with colorful band interdependencies — it’s easy to hear why. Today, working in close partnership with Solid State Logic®, UA proudly unveils the SSL E Series Channel Strip plug-in for UAD-2 — an exacting circuit emulation of this certified hit-making machine.
SSL E Channel Strip Screenshot Figure 141. 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 that knob settings, when compared to the graphical user interface silkscreen numbers, may not match the actual parameter values. This behavior is identical to the original hardware, which we modeled exactly.
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” and “Brown” modes with the EQ Type switch. See “EQ Type” on page 442 for more information. High Pass The left knob determines the cutoff frequency for the high pass filter. Rotate clockwise to reduce low frequencies.
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 441). Compressor/Limiter When UAD SSL E Channel Strip is used in a stereo-in/stereo-out configuration, two separate dynamics processors are active (one for each stereo channel).
This compressor has an automatic make-up gain function. As Threshold is lowered and compression increases (as knob is rotated clockwise), output gain from the module is increased automatically to compensate. Compress Release Release sets the amount of time it takes for gain reduction to cease once the input signal drops below the threshold level. Longer release times can smooth the transition that occurs when the signal dips below the threshold, which is especially useful for material with frequent peaks.
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 hardware consoles. Gate 2 (G2) Gate 2 mode operates the same way as Gate 1, but has a different “no-chatter” response characteristic that is derived from later versions of the hardware. Threshold defines the input level at which expansion or gating occurs. Any signals below this level are processed. Signals above the threshold are unaffected.
Expand Attack Attack defines the duration between the input signal reaching the threshold and processing being applied by the expander/gate. Attack time is normally auto-sensing and program dependent. When Fast Attack is enabled, attack time is 1ms. Fast Attack is active when the “F.ATK” LED (4K: “F.ATT”) is illuminated. To toggle Fast Attack, click the LED or its label text. These three buttons determine the status of the dynamics processors.
EQ 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 Frequency (LF, orange knobs). The high and low bands can be switched from shelving mode into bell (peak/dip) mode. The two midrange bands are fully parametric. The EQ module can be disabled altogether or routed for dynamics sidechain keying. Two different types of SSL EQ are available.
High Frequency (HF) Band HF Gain This control determines the amount by which the frequency value for the band is boosted or attenuated. 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. HF Frequency This control determines the band frequency to be boosted or attenuated by the band Gain setting. The available range is 1.5 kHz to 16 kHz in both Black and Brown modes.
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. Low-Mid Frequency (LMF) Band LMF Gain This control determines the amount by which the frequency value for the band is boosted or attenuated.
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. In Bell (peak/dip) mode, frequencies below and above the frequency value are boosted or attenuated. In Black mode, the LF Bell Q is 1.
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-dynamics is active when the LED below the button is illuminated. Global The vertical LED-style metering provides a visual indication of the signal levels at the input and output of the plug-in (the meters are not calibrated).
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 signal. Usage Notes The SSL E Series channel has been used to mix more hit records than any other in history. Its no-nonsense feature and control set make it easy to get the sound you're looking for.
The compressor's simple control set allows for a wide variety dynamics control, from transparent to aggressive. A fully continuous ratio allows for the full range of knee from very gentle to fully limited. A fixed two position attack and the continuously adjustable release are perfect control sets for general console dynamics control.
CHAPTER 46 SSL G Bus Compressor Large Format Console Dynamics The SSL G Series Bus Compressor plug-in for UAD-2 is an incredibly faithful circuit emulation of the legendary 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 successful studio production console.
SSL G Bus Compressor Controls Threshold Threshold defines the signal level at which the onset of compression occurs. 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 decreased and more compression occurs, output level is typically reduced. Adjust the Make Up control to modify the output to compensate if desired. Make Up Make Up controls the signal level that is output from the plug-in.
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 Ratio defines the amount of gain reduction to be processed by the compressor. For example, a value of 2 (expressed as a 2:1 ratio) reduces the signal above the threshold by half, with an input signal of 20 dB being reduced to 10 dB. The available Ratio values are 2:1, 4:1, and 10:1.
Fade Rate Fade Rate determines the amount of time that will pass between the Fade button 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. Fade times immediately reflect the current Fade Rate value. Therefore a fade out that has already been initiated can be accelerated by changing Fade Rate during the fade out.
gree, and the “Auto” setting provides a program-dependent, multi-stage release for the greatest degree of transparency. Use the 2:1 for the most transparent sound and 10:1 for tougher, more audible sound, or 4:1 for in between. Usually, this processor is meant to be used with minimal gain reduction. In most cases, setting the threshold for 1-2 dB average gain reduction is most common, with occasional transients that go beyond the average. In quieter passages little or no meter movement will occur.
CHAPTER 47 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” analog tape machine — including classics from Metallica, Stevie Wonder, Tom Petty and Jeff Buckley — serve as shining examples of the musicality of analog tape.
Studer A800 Screenshot Figure 143. The Studer A800 plug-in window Operational Overview The Studer A800 for UAD-2 provides all of the original unit’s desirable analog 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. The reel deck IPS control steps through the three tape speed choices available on the original hardware (7.
+6dB, +7.5dB, or +9dB calibration levels, which can be used at their recommended settings, or tweaked for additional tonal options. Input, Sync and Repro 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 global adjustment of any parameters for all Studer A800 instances in your session.
Primary & Secondary Controls The primary controls (those that are typically most used) are on the main panel at the bottom portion of the interface. Additional (typically less used) controls are available on the secondary panel. The secondary panel (see Figure 144) is accessed by clicking the Studer A800 label or the “OPEN” text label above it. For detailed descriptions of the parameters, see “Primary Controls” on page 458 and “Secondary Controls” on page 461. Figure 144.
Ganged Operation The UAD Studer A800 implements a control ganging feature that allows easy simultaneous parameter modification for all instances of the plug-in. The feature enables the DAW to emulate the multitrack tape deck scenario more accurately, where a single change to some multitrack machine parameters (such as tape speed, formula, and calibration settings) would affect all tape channels. See “Gang Controls” on page 463 for details.
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 the most popular 2” magnetic tape formulas are modeled in the A800 plug-in: 250, 456, 900, and GP9. Each type has its own subtle sonic variation, distortion onset, and “tape compression” characteristics.
• 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 462) 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 the recording of specific genres.
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 Output control can be lowered to compensate. Tip: Click the “0” control label text to return to the Input value to 0. Output acts as an outside gain control (like an external console fader) and adjusts the gain at the output of the plug-in. The available range is –24 dB to +12 dB.
Equaliser (Emphasis EQ) The Equaliser buttons determine the active Emphasis EQ values 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. When the Tape Speed is 30 IPS, neither value is available (the LEDs are dimmed) because the EQ is fixed with the AES emphasis curve. When the value is set to NAB, the Hum Noise frequency is 60 Hz (the United States standard).
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. Auto Cal The Studer A800 has individual parameters for Bias, HF Record EQ, and Sync/Repro EQ. On the hardware tape machine, these calibration controls are usually adjusted whenever Tape Type, Tape Speed, or Emphasis EQ is changed.
• 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 to be automatically adjusted for all instantiations.
Figure 145. 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 content into the incoming signal prior to the tape non-linearity. The control provides a continuous “boost filter” gain and affects saturation characteristics. Note: This filter is prior to the tape record circuit, while the other EQs (Sync, Repro) are for tape playback only.
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. Sync EQ and Repro EQ are used as filters to shape the frequency response of the system in maintaining a flat response, but they may be used on their own for high or low frequency adjustment.
Hum Noise The Hum Noise frequency is dependent on the setting of the Equaliser (Emphasis EQ) control (page 462). 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 (Emphasis EQ) LEDs are not illuminated, indicating that the Emphasis EQ is set to AES. However, in 30 IPS mode, the Equaliser switch can still be changed to set the frequency of Hum Noise.
CHAPTER 48 Teletronix LA-2A Leveler Collection History For 50 years, the original and most revered and relied upon transparent optical gain reduction design The Teletronix LA-2A Leveling Amplifier rivals only the Universal Audio 1176 as the most ubiquitous outboard processor on the planet, and is considered the most revered and relied upon vocal compressor in history.
Teletronix LA-2A Leveler Collection Screenshots Figure 146. The Teletronix LA-2A Silver plug-in window Figure 147. The Teletronix LA-2A Gray plug-in window Figure 148. The Teletronix LA-2 plug-in window Figure 149.
LA-2A Plug-In Family The complete LA-2A family is comprised of four individual plug-ins, as seen on the previous page. Each variation has its own unique sonic characteristics. Teletronix LA-2A Leveler Collection The Teletronix LA-2A Leveler Collection (introduced in UAD v6.5) provides access to three historical and highly coveted revisions in the Teletronix product line.
Operational Overview Applications In the 60s and 70s, the LA-2A and 1176 became inextricably linked as the must-have dynamic tools of the day. If the 1176 is to the Stratocaster in terms of immediacy and flexibility, then the LA-2A is to the Gibson Les Paul in terms of warmth and one-of-a-kind, magical sonic distinction. An important characteristic of the T4 photocell response is that it is both program and frequency dependent.
Reference Level Plug-ins in the Teletronix LA-2A Leveler Collection operate at an internal reference level of –12 dBFS. This enables more range in the primary controls (Peak Reduction and Gain) before the I/O distortion characteristics become apparent (signals at the input of these plug-ins can be pushed higher before they distort). For additional information about internal reference levels, see “Operating Levels” in Chapter 7 of the UAD System Manual.
LA-2A Controls Each model in the LA-2A plug-in collection has the same control set. The parameter descriptions below apply to all models unless otherwise noted. Peak Reduction This control sets the amount of signal compression by adjusting the trigger threshold. Increasing the value lowers the threshold, and therefore increases the amount of compression. The available range is 0 dB (fully counter-clockwise) to -40 dB (fully clockwise).
Compress/Limit This switch sets the compression ratio of the leveler. When set to Compress, the compression ratio is approximately 3:1 and when set to Limit, the ratio is approximately infinity:1. However, the compression ratios are nonlinear and frequency dependent, so these figures are not absolute. Note: Like the original hardware, this control is unavailable on the Teletronix LA-2 plug-in. The plug-in is “hardwired” in Limit mode.
Historical Background In the 1950s while at Parsons Electronics, Electrical Engineer Jim Lawrence was quietly asked to join the Titan Missile Program based at Cal Tech's Jet Propulsion Lab and was assigned to develop optical sensors for the program. Fortunately for everyone, the technology developed from Lawrence's work lead back to a more peaceful deployment of the optical sensor, as the detector in his future Leveling Amplifier.
The Teletroniz LA-2A Leveler Collection Original Hardware UAD Powered Plug-Ins Manual - 476 - Chapter 48: Teletronix LA-2A Leveler Collection
CHAPTER 49 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 console 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 making.
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 151). These are unique in that the switches can be pushed in simultaneously, offering distinct cut filter combinations with unusual filter curves.
Trident A-Range EQ Controls Phase Low Pass Filters The Phase (Ø) button inverts the polarity of the signal. The signal is inverted 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 simultaneously in any combination. The available cutoff frequencies are 15 kHz, 12 kHz, and 9 kHz with a slope of 12 dB per octave.
Low-Mid Band The low-mid EQ offers peak/dip “bell” equalization for the middle-to- low frequencies. 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. The available range is approximately ±15 dB. The gain value is zero when the slider is in the center position.
High Pass Filters Three high pass filters are available, and they can be used simultaneously in any combination. The available cutoff frequencies 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” to the others. For example, engaging the 50 Hz filter will rolloff frequencies below 50 Hz, but engaging 100 Hz as well will also attenuate frequencies below 50 Hz, even more than if 50 Hz was used by itself.
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.
CHAPTER 50 UA 1176 Classic Limiter Collection History The Definitive Collection of the World’s Most Famous Compressors The original Universal Audio 1176, designed by audio Renaissance man and UA founder M.T. “Bill” Putnam, represented a major breakthrough in limiter technology. The first true peak limiter to feature all-transistor circuitry and FET gain reduction, the original 1176 offered superior performance and a signature sound — including a lightning-fast 20 microseconds attack time.
UA 1176 Screenshots Figure 153. The UA 1176 Rev A plug-in window Figure 154. The UA 1176LN Rev E plug-in window Figure 155. The UA 1176AE plug-in window Figure 156. The UA 1176LN Legacy plug-in window Figure 157.
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 Collection The UA 1176 Limiter Collection bundle (introduced in UAD v6.2) provides three distinct 1176 revisions, representing over 40 years of design iterations to the original 1176 — the world's most recognized compressor.
UA 1176LN/SE Legacy The 1176 legacy bundle includes the UA 1176LN Legacy and UA 1176SE Legacy plug-ins. These first-generation plug-ins run on UAD-1 and UAD-2 devices. To accommodate the limited DSP resources of the UAD-1, the input transformer 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, vocal, 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 amplifiers, these models can also be used as “tone boxes” to add 1176 color without compression/limiting by disengaging the ratio control (all Ratio buttons “up”).
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. It can add a very useful, gritty compression effect.
1176 Controls Each 1176 plug-in variation has the same control set, so they are only detailed once. The parameter descriptions below apply to all models unless otherwise noted. Input Input adjusts the amount of gain reduction as well as the relative threshold. Rotate the knob clockwise to increase the compression amount. Like the original hardware, the label values are somewhat arbitrary; the knobs are not calibrated to any particular dB values and levels will vary between the various plug-in models.
The behavior of the Attack knob varies slightly between the models, as detailed below. UA 1176AE Attack The 1176AE offers a unique, fixed 10 ms “SLO” Attack mode when this control 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. This enables these models to add 1176 color without dynamics processing.
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 desirable 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 attack- release cycles.
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 release 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. All Button mode is available in all 1176 models.
VU Meter This is a standard VU meter that displays either the amount of gain reduction, or output level, depending upon the setting of the Meter Function switch. 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.
CHAPTER 51 UA 610 Tube Preamp & EQ Collection Historical to Modern Tube Amplification from Universal Audio The Universal Audio 610-A Modular Amplifier was designed by audio renaissance man and UA founder M.T. “Bill” Putnam, and set a major milestone in console design. An all tube and transformer class-A design with feedback style EQ, the 610-A was the first preamp design to include echo sends and modularity which allowed channels to be swapped mid-recording session.
UA 610 Screenshots UA 610-A “Vintage” plug-in UA 610-B “Modern” plug-in Figure 159.
The UA 610 Plug-Ins The UA 610 Tube Preamp Collection is comprised of two distinct plug-ins, as shown in Figure 159 on page 495: The vintage UA 610-A, and the modern UA 610-B. Each plug-in has its own modeling and each captures its unique segment in the 610’s sonic history. The complete signal path is modeled for both 610 plug-ins, including tube amplifiers and transformer components, along with all the phase shift and distortion characteristics that are inherent in each hardware preamp.
Control Differences The UA 610-A and UA 610-B plug-ins (Figure 159 on page 495) have similar controls and layouts. For some parameters, the function of these controls is identical for both plugins. In the parameter descriptions below, controls and/or descriptions that are common to both plug-ins are detailed at the top of each control name section. Any differences and/or controls that are unique to each specific plug-in are detailed after the common description.
Hardware Input Impedance All Apollo mic preamps feature input impedance switching in analog hardware that can be physically switched by Unison plug-ins for physical, microphone-to-preamp resistive interaction. This impedance switching enables Apollo’s preamps to physically match the emulated unit’s input impedance, which can significantly impact the sound of a microphone. Because the electrical loading occurs on input, prior to A/D conversion, the realism is faithful to the original target hardware preamp.
Input Level Controls Input levels for the UA 610 plug-ins are controlled by the overall combination of the input select, pad, and gain controls available within each plug-in. These parameters, which vary per model, control the first tube gain stage in the plug-ins. Generally speaking, higher input gains will produce more color in the signal. Input Select The 610 hardware has both mic and line level inputs. The Input Select control switches between the “virtual input jacks” in the emulated models.
Unison Impedance When UA 610-A or UA 610-B is used in a PREAMP insert within the Apollo Console application, the hardware input impedance of the Apollo mic preamp is switched to match the value in the plug-in for physical, microphone-to-preamp resistive interaction. With UA 610-B, matching the microphone to the closest impedance value is generally recommended, but this parameter can be used creatively and will not harm equipment connected to the Apollo mic preamp.
UA 610-B Input Gain The 5-position rotary Gain switch changes the level at the tube input stage. The control attenuates the input signal by –10 or –5 dB, or adds +5 or +10 dB of gain. In the center “0” position, neither gain nor attenuation is applied. Input Pad Additional attenuation for the Mic input is available via the two-position Pad switch. Setting the switch to the “–15” (up) position attenuates the Mic signal at the tube input stage by –15 dB. In the down position, no attenuation is applied.
EQ The UA 610 preamps feature high and low frequency boost/cut shelving filters with stepped gain controls. The equalizers utilize a feedback-style design which effects the distortion characteristics of the output stage. UA 610-A The low frequency (“L.F.”) shelf EQ has a fixed cutoff frequency of 100 Hz. Lo EQ Gain This rotary switch determines the amount of boost or cut applied to the low frequency signal. Fixed values of ±6 dB or 0 dB can be selected. When set to 0 dB, the filter is inactive.
UA 610-B The high frequency (“HI”) shelf EQ has a selectable cutoff frequency which can be cut or boosted by various amounts. Hi EQ Frequency This switch determines the cutoff frequency (4.5 kHz, 7 kHz, or 10 kHz) of the high shelf EQ. This switch has no effect if the Hi EQ Gain value is zero. Note: Like the hardware, high frequency values are not consecutively ordered. Hi EQ Gain This rotary switch determines the amount of boost or cut applied to the high frequency signal.
One of these 610 desks is the famous Wally Heider “Green Board.” This extremely well-built example was originally fabricated by Frank DeMedio for Wally Heider’s Remote Recording Service in the early ‘60s. Wally originally handled many of the live recording dates booked by Putnam. This console recorded and mixed some of the greatest performances of the era, from many live shows with the “Rat Pack,” to recording the very last concert under the baton of Igor Stravinsky in Los Angeles.
The original Wally Heider “Green Board” console containing 12 vintage UA 610-A preamplifier modules The Universal Audio 2-610 Dual Channel Tube Preamplifier UAD Powered Plug-Ins Manual - 505 - Chapter 51: UA 610 Tube Preamp & EQ Collection
INDEX Numerics Attack Knob 88 1176 Classic Limiter Collection 483 Attenuation Knob 390, 1176 Controls 489 Auto Mode 358 1176 History 483 1176 Plug-In Family 485 1176 Screenshots 484 1176LN 483 4K Buss Compressor 449 4K Channel Strip 435 610 Screenshots 495 610 Tube Preamp & EQ Collection 494 610 Tube Preamp Collection 494 A A/B Selector 72, 271, 342, 414, 416, 422 acoustical space 105 AGC Mode 166 Air Blending 107 Air Density Menu 106 algorithm 104 All Button 373 Allen Sides 290 Ampex ATR-102 18 Amp
INDEX Compressor Output Knob 89 Control Bats 71 Control Grouping 340 Controls Link 166, 171 Cooper Time Cube 80 Cooper Time Cube Controls 81 Cooper Time Cube Screenshot 80 CS-1 Channel Strip 86 Curve Control Bats 71 Curve Control Points 377 Customer Support 17 Cut Enable Button 73 Cut Filter 130 Cut Frequency Knob 74 Cut Type Menu 73 Dual Mode 341, 413 Dynamic Room Modeling 288 Dynamics 275 Dynamics Meters 380 E Early 111 Early & Late Adjustment 111 EL7 FATSO 116 Empirical Labs EL7 FATSO 116 EMT 140 Con
INDEX FATSO Controls 120 I FATSO Functional Overview 117 Input 112, FATSO Jr. 116 Input Mode Switch 416 489 FATSO Screenshots 117 K FATSO Sr.
INDEX Low Frequency Controls 389, Low Frequency Knob 132 Low Gain Knob 133 L-Pan Knob 92, 95 392 M Mac OS X 15 Manley Massive Passive EQ 214 Manual Conventions 15, 17 Massive Passive Controls 217 Massive Passive EQ 214 Massive Passive Latency 224 Massive Passive Mastering EQ 216 Massive Passive Notes 224 Master 72 Master Level Knob 72 Materials Blending 107 Materials Blending Bars 107 Materials Menus 106 Materials Panel 105 Materials panel 101 Meter 493 Meter Pop-up Menu 89 Microphones 290 Mix 113, 367 Mi
INDEX Output Knob 88, 92, 95 Output VU Meter 134 Overview 86, 96, 178, 184, 244, 249, 256, 262, 273, 320, 326, 330, 335, 364, 370, 417, 424 P Parameter Copy Buttons 341 Parametric EQ 75 Parametric Type Selector 75 Peak 185 Peak Level 245, 250, 257, 418 Peak Reduction 473 Plate 140 Overview 129 Plate Select Switch 131 Platforms 15 Positioning panel 101, 110 Power 414, 416 Power Lamp 135 Power Switch 135, 342 Precision Buss Compressor 96, 320 Precision Limiter Meters 358 Precision Limiter Screenshot 357 Pr
INDEX Resonance panel 101 Response Curve Color 70 Response Curve Display 70 Reverb Time Meters 131 Reverberation panel 101, 109 Roland 415, 416 Roland CE-1 Controls 412 Roland CE-1 Overview 411 Roland Dimension D 415 Roland Dimension D Controls 416 Roland Dimension D Overview 415 Roland Dimension D Screenshot 415 Roland RE-201 417 Roland RE-201 Controls 418 Roland RE-201 Interface 418 Roland RE-201 Screenshot 418 Room Shape and Material 396 R-Pan Knob 92, 95 RS-1 Controls 94 RS-1 Reflection Engine 93 Spati
INDEX U UA 1176 Classic Limiter Collection 483 UA 1176 Limiter Collection 485 UA 1176 Screenshots 484 UA 610 Screenshots 495 UA 610 Tube Preamp Collection 494 UA 610-B Controls 499 UAD Nomenclature 15 Users Forum 16 V Video Documentation 16 VU Meter 418 W Welcome 14 Wet 113 Wet Solo Button 134 Wet/Dry Mix Knob 92, Width 328 Width Knob 134 Windows 15 www 16 95 Z Zoom Buttons 71 UAD Powered Plug-Ins Manual - 512 - Index
