UAD POWERED PLUG-INS USER GUIDE SOFTWARE VERSION 6.1 MANUAL VERSION 111025 Universal Audio, Inc. 1700 Green Hills Road Scotts Valley, CA 95066-4926 Voice: +1-831-440-1176 Fax: +1-831-461-1550 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.
Universal Audio, Inc. End User License Agreement By installing the software, you confirm your acceptance of the Universal Audio and third-party End User License Agreements, as well as the Universal Audio terms of service and privacy policy, all of which can be found at: http://www.uaudio.com/eula This Agreement is between Universal Audio, Inc., and you. IMPORTANT PLEASE READ THIS LICENSE AGREEMENT CAREFULLY BEFORE INSTALLING THIS SOFTWARE.
7. Termination. To the extent permitted by law, and without prejudice to any other rights Universal Audio may have, Universal Audio may terminate your license if you materially breach these terms and conditions. Upon termination by Universal Audio, you will return to Universal Audio, at your expense, the Software, including documentation, and any copies thereof. 8. United States Government Rights. The Software and Documentation are provided with RESTRICTED RIGHTS.
TABLE OF CONTENTS Chapter 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 18 Welcome!. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 18 Features . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 20 The UAD System . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
TABLE OF CONTENTS Chapter 5. Using Multiple UAD Devices. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 55 Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 55 Plug-In License Policy . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 55 UAD Link Licensing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
TABLE OF CONTENTS Range Limits . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 98 Entering Values . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 98 Out of range . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 99 Modes with Tempo Sync . . . . . . . . . . . . . . . . .
TABLE OF CONTENTS Primary Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 151 Secondary Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 159 Manual Calibration Procedure . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 167 Manual Calibration Notes . . . . . . . . . . . . . . . . . . . . . . . . . .
TABLE OF CONTENTS Chapter 17. DreamVerb . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 203 Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 203 Signal Flow . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 204 Resonance (Equalization) Panel . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
TABLE OF CONTENTS Webzine Article. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 253 Chapter 21. EP-34 Classic Tape Echo. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 255 EP-34 Overview. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 255 EP-34 Tape Echo Screenshot . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
TABLE OF CONTENTS Chapter 26. LA-3A Compressor . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 290 Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 290 LA-3A Screenshot . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 290 LA-3A Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
TABLE OF CONTENTS Moog Filter Screenshot . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 333 Moog Filter Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 333 Moog Filter SE . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 339 Moog Filter Latency . . . . . . . . . . . . . . . . . . . . . . . . .
TABLE OF CONTENTS Other Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 371 Neve 33609SE . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 376 Neve 33609 Latency . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 376 Chapter 37. Neve 88RS Channel Strip . . . . . . . . . . . . . . . . .
TABLE OF CONTENTS Precision De-Esser Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 428 Operating Tips . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 430 Chapter 41. Precision Enhancer Hz . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 431 Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
TABLE OF CONTENTS Precision Multiband Interface . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 462 Band Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 463 Band Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 464 EQ Display . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
TABLE OF CONTENTS Chapter 50. Roland Dimension D . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 503 Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 503 Roland Dimension D Screenshot . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 503 Roland Dimension D Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
TABLE OF CONTENTS Chapter 55. Studer A800 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 542 Multichannel Tape Recorder . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 542 Studer A800 Screenshot . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 543 Operational Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
CHAPTER 1 Introduction Welcome! Congratulations, and welcome to the UA Family! You now own the best sounding, most powerful sonic upgrade available for your DAW. The UAD-2 and version 6 software represents the culmination of a multi-year development effort delivering the latest in state-of-the-art audio processing technology and is the next evolution of the revolutionary, award-winning UAD Powered Plug-Ins & DSP Device Platform.
The UAD-2 also includes version 6 of the UAD software, which has major driver enhancements and an all-new UAD Meter & Control Panel. Version 6 features multicore-aware drivers, LiveTrack™ for low-latency tracking and monitoring through UAD-2 plug-ins, and L.O.D.E.™ [Live Optimizing DSP Engine], which dynamically balances the load on the UAD-2.
Features • High-bandwidth x1 PCI Express card (PCIe 2.0 compatible) • UAD-2 Solo: low-profile, half-height, short PCIe card • UAD-2 Duo/Quad: full-height, short PCIe card • UAD-2 Satellite: Duo or Quad in FireWire enclosure w/power supply • 1, 2 or 4 Analog Devices SHARC floating-point processors • UAD-2 averages 2.5X(Solo), 5X(Duo), 10X(Quad) the power of the UAD-1 • Multi-device support for up to four UAD-2’s in one computer • Cross-Platform* for Mac OS X 10.5/10.6/10.
The UAD System The UAD Powered Plug-Ins package is a hardware-plus-software system that consists of one or more UAD DSP accelerator devices combined with the Powered Plug-Ins software. Multiple UAD package types are available in the product line to accommodate your particular hardware system, processing needs, and budget. The difference between each package are the UAD device type and the selection of plug-ins that are bundled with the device.
User Manuals This complete user manual for the product (the document you are reading now) is included in the software bundle. The filename of the manual is UADManual.pdf. The User Manual is the primary product user documentation. It is on the software CD-ROM, and is also placed inside the Powered Plug-Ins Documentation folder on the hard drive during software installation. Direct Developers As of version 6.0, UAD Powered Plug-Ins includes plug-ins from our Direct Developer partners.
Screen Shots Screenshots in this manual may be taken from the Windows and/or Mac version of the software, and are used interchangeably when the content and functionality of the screenshot is the same on both platforms. Slight variations in the appearance of a screenshot between operating systems are inevitable. When the content of and function of the software represented in a screenshot is identical on both platforms, no differentiation is made in the screenshot title.
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. Phone Support Customer Service & Technical Support USA toll-free: 877-MY-UAUDIO (1-877-698-2834) International: +1-831-440-1176 FAX: +1-831-461-1550 Online Support To request online support, please visit our support page, then click the “Submit Support Ticket” button to create a help ticket: • http://www.
CHAPTER 2 UAD Installation Overview UAD Powered Plug-Ins installation and configuration consists of four steps: • UAD software installation Insert the enclosed CD-ROM and run the installer. • UAD hardware installation Follow the instructions in this chapter. • UAD device registration Add your device to your my.uaudio.com Account. • UAD plug-in authorization Download and apply UAD authorization file. UAD vouchers can be redeemed and optional plug-in licenses can be purchased anytime.
System Requirements UAD Powered Plug-Ins require the following hardware and software: Windows (x86 and x64) • Microsoft Windows XP (Home, Professional, or x64 Edition), Windows Server 2003, Windows Vista, or Windows 7 Mac (32-bit and 64-bit*) • Mac OS X 10.5 Leopard, 10.6 Snow Leopard, or 10.
We test the specific host applications listed in Table 1 on page 27 for compatibility with UAD Powered Plug-Ins, and only these hosts are supported by our technical staff. We recommend the latest versions of these host applications for optimum performance. Host applications that are not listed may work with UAD Powered Plug-ins, but compatibility with them cannot be guaranteed.
UAD Software Installation Install Software First If you are installing UAD Powered Plug-Ins for the first time, install the software before installing the UAD device(s). This is particularly important on Windows systems. If you are updating to a newer version of the software or installing additional UAD devices, it is not necessary to remove the previous UAD software or hardware from the system, but you should still install the newer software before adding new devices.
3. When installation has completed, power down your system to install the UAD card(s) following the instructions on page 30. If the UAD device is already installed or you are updating, restart the system. 4. After restarting, the New Hardware Wizard will try to locate the new drivers. Follow these steps: • a) Select the “Install the software automatically (Recommended)” option, and click “Next.” • b) The UAD device driver will be installed, click “Finish” when it is complete.
Verify Install You can use the UAD Meter & Control Panel to verify installation (see “Verifying Installation” on page 35.
However, if a computer system has both PCI/PCI-X and PCIe slots, the UAD–1, UAD–1e, and UAD-2 can all be installed and used simultaneously as a multidevice system. See Chapter 5, “Using Multiple UAD Devices.” Important: The UAD-1 will ONLY work in PCI and PCI-X slots, and the UAD1e, UAD-2, and UAD-Xtenda will ONLY work in PCI Express (PCIe) slots. Be extremely careful to only install the UAD device(s) into a compatible slot! PCI/PCI-X: UAD-1 only PCI Express (PCIe): UAD-1e UAD-2 UAD-Xtenda Figure 1.
5. Before handling the UAD card, discharge any static electricity by touching the outer casing of the power supply. 6. Remove the UAD card from its protective anti-static bag. Do not touch the gold edge connector contacts. 7. Hold the card gently by the top edges, and line up its connector with the slot inside the computer. Figure 2. UAD-1 card installation in PCI slot. UAD-2 in PCIe is similar. 8. When the connector and slot are aligned, press the card into the slot with firm, even pressure.
Important: All UAD plug-ins must be authorized before they can be used (unauthorized plug-ins can be used in demo mode for 14 days without authorization). Authorization is accomplished by clicking the “Authorize Plug-Ins” button in the UAD Meter & Control Panel application. The procedure is in the next section. Authorize Plug-Ins Procedure This procedure provides only the step-by-step process for UAD authorization. A complete overview of UAD authorization, my.uaudio.
Figure 3. Double-click the downloaded UAD authorization file. Offline Authorization If the UAD computer is not connected to the internet, you will need to manually transfer the registration URL into the browser of a web-enabled computer. To download and apply the UAD authorization file if not connected to the internet: 1. Install UAD version 5 (or higher) software and the UAD hardware using the procedures detailed in this chapter (the system must be operating properly). 2.
7. The authorization file should begin downloading automatically. Click the “Download Authorization” link to download the auth file if not. 8. Transfer the authorization file to the UAD computer. 9. Double-click the authorization file. The authorization is loaded onto the UAD device(s) and after a few seconds the “Authorizations Updated Successfully” window appears (Figure 3). Online UAD authorization is now complete, and UAD Powered Plug-Ins are ready for use. 1.
To confirm proper UAD installation and operation: 1. Launch 2. Open the UAD Meter & Control Panel application (see page 62). the System Information panel (see page 64). 3. When the UAD device is displayed in the Hardware area and its status is “OK” then the UAD driver is communicating with the UAD device and the system is operating properly. Done! UAD-2 LED The UAD-2 has an LED indicator. This LED provides certain indications about the operational state of the UAD-2.
Learn More After installation, registration, and authorization, you’re ready to use UAD Powered Plug-Ins. We especially recommend reading chapters 3, 4, 6, and 7 in this manual to gain important insights about using the product. Visit our support site for the latest technical information and additional information not included with the software, or to contact technical support about any technical difficulties: • http://www.uaudio.
CHAPTER 3 UAD System Overview The UAD Environment The complete UAD Powered Plug-Ins environment consists of many components. These components are: • One or more UAD DSP hardware accelerator devices • The UAD device drivers • The UAD plug-ins • The host computer system • The plug-in host application software • An audio I/O interface and its drivers All these components operate together simultaneously within one computer to form the complete audio processing system.
Note: Multiple UAD hardware types can be used concurrently in the same computer. For details, see Chapter 5, “Using Multiple UAD Devices.” UAD-2 is our second-generation UAD product line that offers many improvements over the original UAD-1, most notable is significantly increased DSP horsepower. UAD-2 Family A variety of UAD-2 models are available to suit various processing and protocol needs.
The UAD-2 SOLO/Laptop device (and UAD–Xpander) can be used with an optional 3rd-party PCIe-to-ExpressCard adapter card, which enables the UAD ExpressCard to be used in desktop computers that have PCIe expansion slots. This system provides a convenient way to switch the UAD-2 SOLO/Laptop between desktop and notebook computers. UAD-2 Satellite The UAD-2 Satellite is available in Duo and Quad versions and is enclosed in an attractive housing that uses an external power supply.
UAD-Xpander The UAD-Xpander is a UAD-1e card housed in an external chassis that interfaces to the computer using an ExpressCard expansion slot. UAD-Xpander has special instructions for installation, connectivity, and use. See Chapter 10, “UAD ExpressCard Products” for complete details. UAD-Xtenda The UAD-Xtenda is an optional ExpressCard-to-PCIe adapter card that enables the UAD-Xpander (and UAD-2 SOLO/Laptop) to be used in desktop computers that have PCIe expansion slots.
The UAD Software The UAD Powered Plug-Ins software consists of the UAD installer, the UAD plug-ins, the UAD Meter & Control Panel application, the UAD device drivers, and the user documentation. Installer The UAD Powered Plug-Ins software installer contains all the UAD software components (plug-ins, drivers, and documentation) and copies them to disk during installation. There is one UAD software installer for each OS platform (Windows and Mac).
The UAD Meter & Control Panel (page 61) is a utility application that is used to monitor and configure UAD device resources. It has two main windows: the UAD Meter, and the UAD Control Panel. UAD Meter & Control Panel The UAD Meter window (Figure 7 on page 61) displays the current DSP, program, and memory status of the UAD hardware in realtime. The UAD Control Panel window (page 64) has multiple panels that display and modify various system, plug-in, and global configuration parameters.
The DAW Environment The digital audio workstation (“DAW”) environment is an overall system where audio production takes place, either with or without UAD Powered Plug-Ins. The UAD system is an add-on to the computer-based DAW that can dramatically improve the sonic quality and performance of the DAW, while still working entirely from within the DAW.
Each host application has its own set of system requirements. In addition to the UAD-specific system requirements, the host computer must meet the host application system requirements. Because the host application environment is integral to the UAD system but is completely out of our control, we cannot claim compatibility with all hosts that may support use of the UAD plug-in formats. See “Supported Hosts” on page 26 for complete details.
I/O Drivers The audio interface is a hardware device that requires a software device driver for the host computer OS. In addition to instructing the OS on how to control the interface hardware, the driver acts as a software “middleman” between the host application and the I/O ports. The host application uses the driver to access the I/O for signal routings.
CHAPTER 4 My.uaudio.com My.uaudio.com is the Universal Audio online store where UAD devices are registered to your account and UAD plug-in authorizations are obtained. Bundled plug-in vouchers are redeemed at the store and optional plug-in licenses can be purchased at my.uaudio as well. This chapter provides details about UAD plug-in authorization and licensing, and the my.uaudio.com store. Important: All UAD plug-ins must be authorized before they can be used.
Bundled Plug-Ins UAD plug-ins that are bundled with the UAD device are automatically licensed when the device is registered. After registration, the Authorization File must be loaded. Note: UAD devices must be authorized to run bundled (included) plug-ins. See “Authorization Overview” on page 48 for details. Vouchers Vouchers included in UAD retail packages have dollar values that are as “good as cash” for buying UAD plug-ins at my.uaudio.com. Vouchers are applied to your my.uaudio.
Video Help Our support website contains many helpful videos that explain how to register and authorize the product, obtain optional plug-ins, and more: • http://www.uaudio.com/support/uad/videos.html Procedure This section is a detailed overview of the entire UAD authorization system. For the step-by-step authorization instructions, see “Authorize Plug-Ins Procedure” on page 33 in Chapter 2, “UAD Installation.
My Systems The UAD devices and plug-ins that you own and can authorize can be found in your My Systems page at my.uaudio.com. My Systems keeps track of which UAD devices are in which computer, and the authorization status of each UAD plug-in. My Systems is managed automatically by my.uaudio.com and the UAD software. When “Authorize Plug-Ins” is clicked, devices in the host system are added to My Systems. When an authorization file is downloaded, it reflects the current state of My Systems.
• The .uad/.uad2 authorization file contains plug-in license keys for the specific UAD devices in each computer group at My Systems. The file will not authorize any other UAD devices. • An authorization file is not associated to the host computer or the UAD installer in any way. Instead, it is associated with the UAD devices at My Systems (it’s the UAD hardware that is authorized, not the UAD plug-ins or computer).
4. Click the Start Demo button for the desired UAD plug-in. A confirmation window appears and the timed demo can be activated or demo activation can be cancelled. The timed demo is activated for 14 days. Demo Notes • (UAD-2 only) In addition to the Plug-Ins panel, Demo mode can also be activated directly from the UAD Toolbar (see page 87). • (UAD-1 only) Demos cannot be started if any UAD-1 plug-ins are running. If they are, a message instructing you to quit the host application is displayed.
Registration Registration at my.uaudio.com is required for all UAD users to obtain UAD device authorizations, redeem vouchers, purchase optional UAD plug-in licenses, get software updates, and receive customer support. It is also an opportunity for you to inform us of your system details so we can serve you better, and for us to keep you informed about important updates and new product announcements. Account A my.uaudio.
Lots of product information such as features, specifications, audio/video demos, and reviews are available on the plug-in product pages at the store to help you make your buying decisions. Purchase Procedure To purchase and authorize an optional UAD plug-in: 1. Login to your Account at my.uaudio.com. Your email address is your login. 2. Click the “Products” menu and browse the available plug-in selections. 3. Add the items you whish to purchase to your shopping cart. 4.
CHAPTER 5 Using Multiple UAD Devices Overview Multiple UAD devices can be used in a single host computer system for increased DSP capacity. The UAD devices in the multidevice system can be mixed and/or matched in any combination (see “The UAD Hardware” on page 38). When multiple UAD devices are installed in the host computer, the DSP and memory load of the devices are automatically balanced dynamically as UAD plug-ins are loaded and unloaded.
However, as an example, if you have four UAD devices but run a dual-device system in two separate computers, you are required to purchase separate licenses for the second system. “UAD Link Licensing” on page 56 details the exception to this policy, where licenses for portable UAD devices can be linked to a desktop system.
Authorizing Multiple Devices When you buy an optional plug-in, your purchased license is valid for all the UAD devices in the computer. The license is tied to the specific UAD devices in the System (see page 50) when the license was purchased. For example, if you have two devices when you purchase a plug-in, your authorization file will authorize both devices.
The UAD device power requirements are listed in Table 2 on page 58. These figures are measurements of the actual power draw for these devices when running at full DSP load. Note: UAD-2 Satellite cannot be powered by the Firewire bus; it requires the included external power supply adapter for operation. Important: If insufficient power is available to UAD devices, unpredictable behavior may result. Table 2.
Device Info Display UAD DSP and memory resources used for each installed device, and the ability to enable/disable individual devices, is displayed in the System Information window. See “System Information Panel” on page 65 for complete details. Disabling Devices Individual UAD devices can be disabled using the Device Enabled function (see page 67). This can be useful, for example, if creating a session on a system with multiple devices that will be transferred to a system with fewer devices.
For example, if you are trying to minimize latency during tracking by using a smaller buffer size (which will increase host CPU) and need a bit more host CPU, disabling one or more UAD devices during tracking may give the extra pinch of host CPU you need. The buffer size can then be increased and the UAD device(s) re-enabled for mixing.
CHAPTER 6 UAD Meter & Control Panel Overview The UAD Meter & Control Panel application is used to monitor UAD device resources, confirm the UAD system is operating properly, and configure the global UAD Powered Plug-Ins system parameters. The UAD Meter window displays the current DSP, program, and memory status of the UAD hardware in realtime. Figure 7 shows how the UAD Meter appears on when UAD-1 and UAD-2 devices are installed together in the same system.
Launching the UAD Meter & Control Panel Application Windows To launch the UAD Meter in Windows: 1. Double-click the UAD Meter & Control Panel shortcut that was placed on the Desktop during installation. OR, 2. Access the application from the Start Menu at Programs/UAD Powered Plug-Ins/UAD Meter & Control Panel. OR, 3. Double-click the executable file on the hard drive located at C:Program Files/Universal Audio/Powered Plug-Ins/UADPerfMon.exe. Mac To launch the UAD Meter in Mac OS X: 1.
UAD Meter Elements Refer to Figure 7 on page 61 for these element descriptions. Title Bar The title bar at the top of the UAD Meter window contains buttons to quit the UAD Meter & Control panel application and minimize the UAD Meter window, and a drop-menu to access the Control Panel window functions. The Disable Current button disables all UAD Powered Plug-Ins that are currently running.
DSP The DSP gauges display the amount of digital signal processing resources that are being used by the UAD Powered Plug-Ins algorithms. DSP is the primary hardware “juice” that powers the UAD Powered Plug-Ins algorithms. When UAD plug-ins are disabled, DSP requirements are decreased. Program This resource shows how much “program memory” is in use by the UAD-2. Program memory is an on-chip memory that is specific to the UAD-2 DSP processor(s) and is used for certain UAD plug-in resources.
Click the Menu button Figure 8. Accessing the UAD Control Panel Once the main UAD Control Panel window is open, the individual window panels are accessed by clicking the title buttons (Figure 9). Figure 9. The Control Panel window panel buttons Note: The FireWire button is only displayed if UAD-2 Satellite is connected. System Information Panel The System Information panel (Figure 10 on page 66) displays detailed UAD software and hardware information and contains several useful buttons.
Figure 10. The UAD System Information panel Plug-In Latency Section UAD plug-in latency for each device type (UAD-2 and UAD-1) is displayed here. The calculation is based on the audio I/O interface buffer size and the sample rate. The window uses this information to calculate and display the resulting latency in samples and milliseconds. The buffering methods used by UAD-2 and UAD-1 drivers are different for the two device types, which is why the latency differs between the devices.
UAD-2: The displayed latency is the audio I/O interface buffer size when Extra Buffering (page 74) is off. When Extra Buffering is on, the displayed latency is the audio I/O interface buffer size plus 64 samples. UAD-1: The displayed latency is twice the audio I/O interface buffer size. Note: The displayed sample latency number does not include the extra samples produced by upsampled UAD plug-ins.
DSP Load If more than one UAD DSP is installed, information for each of the processors is displayed. DSP, Program (“PGM”), and Memory (“MEM”) loads are displayed as a percentage of total available load for that processor. The number of UAD plug-ins (“PLG”) running on each individual processor is also displayed (not available for UAD-1). Save Detailed System Profile Clicking this button will generate a detailed system profile and prompt you for a location to save the text file to disk.
Plug-In Column All currently installed UAD Powered Plug-Ins are displayed in the Plug-In column. Use the scrollbar to view the entire list if necessary. Status Column The Status column reflects the current state of each UAD plug-in. The status depends on which device (UAD-2 or UAD-1) the plug-in is running on; the status may change if the “Run On” device is changed (page 71).
Figure 11. The UAD Plug-Ins panel Run On Column The Run On column indicates which device (UAD-2 or UAD-1) the specified plug-in will load onto. When both UAD-2 and UAD-1 are installed, you can select between them with the Run On menu in order to better manage your DSP resources.
For example, you could run low-DSP plug-ins such as the UAD CS-1 on the UAD-1, while specifying that resource-hungry plug-ins such as Neve 33609 run on the UAD-2. The Run On column contains a drop menu for each plug-in. Specify which device type (UAD-2 or UAD-1) the plug-in will load onto with the Run On drop menu. Click the disclosure triangle to see the menu, then drag and release on the desired device type.
Figure 12. The UAD Configuration panel DSP Load Limiting Overview Without UAD Powered Plug-Ins installed, overloading the host system with native (host based) plug-ins can cause dropouts and possibly system lockup. Steinberg hosts, for example, provide a switch that allows you to trade latency for stability when the system is overloaded. Similarly, the UAD DSP load cannot exceed 100% without unpredictable behavior.
With the Limit DSP Load feature, the UAD DSP can also be limited so the load cannot exceed 100%, thereby increasing overall system stability in high load situations. With very heavy UAD loads, DSP load limiting may also improve host CPU performance. There are many variables that affect DSP load (sample rate, buffer size, parameter values, mono/stereo, automation, host system, etc). Although these variables are taken into account, the resulting measurement cannot be absolutely accurate.
UAD-2 DSP LoadLock DSP LoadLock (UAD-2 only) reserves the maximum UAD-2 DSP load required by each plug-in, even if certain plug-in features are disabled. This ensures there will always be enough DSP if those features are later enabled, or when automating parameters that affect the DSP load.
AMD-8131 Mode If your computer uses the AMD-8131 PCI controller chipset, check the “AMD-8131 Compatible” box. This will improve UAD performance on these systems. For the new setting to take affect, you must reset the audio interface using one of the following methods: •Close the re-open the session •Stop then restart the audio engine •Modify or reset the audio device settings AMD-8131 Compatible Mode is only required when the device is attached directly to an AMD-8131 PCI bus.
Windows Sonar Compatibility Mode Tick the Sonar Compatibility Mode checkbox to improve UAD-1 plug-in performance when used with Cakewalk Sonar. This mode should be disabled when using different hosts, otherwise audio degradation could occur. Mac Release all DSP resources on Audio Unit bypass Some Audio Unit hosts dynamically bypass plug-ins when they are not being used during playback, for example when no audio is present at the current playback position.
On multi-processor machines Live Mode forces all plug-ins on tracks with UAD plug-ins (and any aux/buses fed by those tracks) to run on one processor only. To allow host-based (native) plug-ins to run on all processors under these conditions, deactivate this setting. Note: Changes to this setting do not take effect until all open sessions containing UAD plug-ins are reloaded. User Interface Settings Controls Mode This setting determines how UAD Powered Plug-In parameter knobs respond to adjustment.
Note: When Use Host Mode is checked, the UAD Meter user interface settings have no effect unless control mode is NOT supported by the host. Mac Toggle initial value modifier Holding the specified modifier key while clicking a parameter control will toggle between the value of the control when the editor was opened and its last edited value. This feature is not supported under Windows. This parameter also affects the “select + click” modifier.
For more information about FireWire and using it with UAD-2 Satellite, see “FireWire Basics” on page 137, “FireWire Bandwidth vs. UAD DSP” on page 143, and “FireWire Bus Power” on page 144. Figure 13. The UAD FireWire panel Current Link Speed The current speed of the FireWire link between the host computer and the UAD-2 Satellite is displayed here. The active link speed can be changed with the Target Link Speed control if FW400 and FW800 are both supported by the host system.
If the host computer has multiple FireWire buses (this is rare), up to two speeds will be displayed, with the values separated by a comma. Note: If the host computer supports FireWire 800 and the Target Link Speed is set to 800, but the Current Link Speed still displays 400, check the bus ordering of the FireWire devices. See “Mixing FireWire Speeds” on page 140 for information on how to properly combine FireWire 400 and FireWire 800 devices on the same bus.
It is important to note that all devices on the FireWire bus share the available data bandwidth of the bus, including hard drives and audio interfaces as well as UAD-2 Satellite. Therefore all of the bandwidth cannot be allocated for UAD use if FireWire hard drives and/or audio interfaces are on the same bus. See “FireWire Bandwidth” on page 141 for additional information. To determine the proper value for the UAD Bandwidth Allocation: 1.
When a UAD plug-in is loaded on UAD-2 Satellite by the host software, the Plug-In Calculator is automatically adjusted, using the session sample rate, Current Link Speed, and UAD Bandwidth Allocation values. Max Stereo Plug-Ins The maximum number of stereo UAD plug-ins that can run on UAD-2 Satellite, based upon the FireWire System settings and a given session sample rate, is displayed here.
Used by other devices This number displays the percentage of FireWire bus bandwidth that is used by all the FireWire devices connected to the bus EXCEPT for UAD-2 Satellite, such as hard drives and audio interfaces. The number is used to help determine an optimum value for the UAD Bandwidth Allocation parameter. FireWire Gauge The FireWire Gauge (Figure 14) reflects the FireWire bus bandwidth currently used by UAD-2 Satellite and all other FireWire peripheral devices (e.g.
Help & Support Panel The Help & Support panel (Figure 15) contains numerous helpful weblinks to help you get the most out your UAD Powered Plug-Ins experience. The button names are self-explanatory. Click a button to launch the URL in your default web browser (you must be connected to the internet to launch the web pages). If the UAD computer isn’t online, much of the information is in the manual you’re reading now; find it in Table of Contents, Index, or search.
CHAPTER 7 Using UAD Powered Plug-Ins Overview Once the UAD device(s) and Powered Plug-Ins have been properly installed, the UAD Powered Plug-Ins are accessed and used just like any host-based plug-in. All UAD Powered Plug-Ins can run concurrently with each other and with host-based plug-ins simultaneously, in any combination.
Logic Pro Figure 17. Launching a UAD Powered Plug-In in Apple Logic Pro The UAD Plug-In Window A typical UAD Powered Plug-In window is shown in Figure 18. The graphical user interface (GUI) typically contains several control parameters, for modifying the behavior of the plug-in, and display elements such as meters, for visual feedback. The UAD Toolbar is also displayed.
Controls View Some host applications have an alternate plug-in display mode feature called Controls View, Parameter Mode, or similar nomenclature. This mode typically displays the control parameters as a list with simple sliders, switches, and menus (the appearance is controlled by the host), which some users prefer. Consult the host documentation for how to display this view. Figure 19.
The Settings menu provides a convenient way to manage your UAD plug-in parameter settings. To select the functions, click the folder icon in the Toolbar, then select an action from the drop-menu that appears. Settings Menu The Settings load/save feature supports presets but not banks. To load and save banks, use the host’s bank management feature (if available).
Windows The default preset location is inside the Presets folder, which is created within the folder specified during software installation. For example, if the default location was selected when running the installer, the location would be: • C:\Program Files\Universal Audio\Powered Plugins\Presets Note: This default location can be changed during installation; the new location will be remembered as the default.
Increased Host Loading using LiveTrack When LiveTrack is active, host CPU loading increases. The host load is directly proportional to the DSP load of the UAD plug-in(s) in LiveTrack mode, however host CPU is never used for Powered Plug-in processing. Extra Latency using LiveTrack When extra buffering is enabled (page 74) or when upsampled UAD plug-ins (page 108) are used with LiveTrack, latency is increased.
Disabled When “DISABLED” is displayed in red, the plug-in will not process audio. This can occur in two situations; either the demo period has expired (click the Buy button!), or the UAD DSP is overloaded and there are not enough device resources to run the plug-in. Status Blank When no text is displayed in the Status area, the plug-in is fully authorized (licensed) and operating normally. In this state, the Buy button is not displayed.
Text Entry Parameter values can be modified directly with text entry. To enter a parameter value using text entry, single-click the parameter value text. The text value will highlight, indicating it is ready to receive a new value. Type in a new value, then press Return, Enter, or Tab, or click outside of the text box. Press Esc if you want to revert to the prior setting without entering the new value. Values entered via text entry are rounded to the closest significant digit.
Table 3.
UAD Devices with UAD-2 Satellite Because Firewire has significantly less bandwidth than the PCIe protocol, if UAD-2 Satellite is running with UAD-2 PCIe card(s) in a multidevice system, UAD plug-ins will load on the UAD-2 card(s) before the Firewire device in order to maximize Firewire bandwidth. See “FireWire Bandwidth vs. UAD DSP” on page 143 for more information UAD-2 Satellite loading.
External MIDI Control UAD Powered Plug-In parameters can be controlled by an external MIDI controller, if this feature is supported by the host application. Each host application has its own particular methods for external MIDI control. Consult the host application documentation for specific instructions on using external MIDI control with the application.
CHAPTER 8 Tempo Sync Overview The time-based parameters of several UAD Powered Plug-Ins can be synchronized to the tempo of the host application using the Tempo Sync feature. When Tempo Sync is activated, the time-based parameters that are available for synchronization are changed to note duration values, and will sync to the tempo of the host application using the displayed note value. Figure 21. The Tempo Sync feature within UAD DM-1L Note: Not all host applications support Tempo Sync.
Sync Activation To activate Tempo Sync, click the “Sync” button within the plug-in interface. The Sync button “LED” will illuminate and the time parameters will change from a time-based display to a note value (see Figure 21). Note: When Tempo Sync is activated, the plug-in will automatically switch the time or rate parameter(s) to the nearest available note value(s) given the range of the parameter in question and the current tempo.
Range Limits Some parameters in Tempo Sync mode cannot access the entire note value range in Table 4, because their maximum values would always be out of range above certain note values (assuming a maximum usable tempo of 300 BPM; 250BPM for Roland RE-201).
Similarly, if 1/12 is entered with text entry, 1/8T is displayed because an eighth note triplet is equivalent to one-twelfth of a measure (if in 4/4 time). Out of range When a parameter note value is out of range of the current tempo note value, the note value is displayed in parentheses on a red background (Figure 22). Figure 22.
Roland RE-201 Sync When the RE-201is in Tempo Sync mode, note values can be imprecise due to the fixed tape head relationships. Values that are imprecise approximations (but are within the available delay time range) are displayed with a “+” or “–” symbol. The leading head in the current mode is accurately synced; the other values are based on the fixed tape head relationship.
CHAPTER 9 UAD Delay Compensation Latency & Delay Compensation When UAD Powered Plug-Ins are used, audio data to be processed by a Powered Plug-In is sent by the host application to the UAD device. The audio is then processed by the UAD device and sent back to the host application. This back-and-forth shuffling of audio data is called “buffering” and it produces a latency (delay) in the audio signal being processed. Latency is inherent in digital audio systems and it can be detected in certain situations.
Table 6. Host Application plug-in delay compensation Implementations Full PDC Platform Partial PDC Platform Steinberg Cubase SX 2 Win / Mac Logic 5, 6, 7.
Host Application Settings For optimum results, the “Plug-in Delay Compensation” option setting should be enabled in the host application. This will provide automatic latency compensation when UAD plug-ins are used on track inserts (and sends/groups/buses if full compensation is supported), so the UAD DelayComp will not have to be used. This option is usually found in the audio or plug-in preferences window.
The UAD-1 Delay Compensator acts as a dummy UAD Powered Plug-In, automatically introducing the necessary amount of latency for tracks which are NOT processed by UAD Powered Plug-Ins. It requires no DSP from the UAD device and allows you specify the number of UAD Powered Plug-Ins instances you wish to compensate. Note: UAD DelayComp is available for UAD-1 only.
Grouping Tracks Requiring DelayComp The UAD DelayComp plug-in is generally used on track inserts. However, when many tracks require delay compensation, instead of placing individual Delay Compensator plug-ins on each track you may find it easier to send the output of each unprocessed tracks to a bus or group. Then simply put one UAD-1 Delay Compensator on that bus or group.
UAD-1 Track Advance Overview The previous discussions on delay compensation (see “Latency & Delay Compensation” on page 101) apply mainly when using only audio tracks. When MIDI tracks are played simultaneously alongside audio tracks, a different (but related) synchronization issue can arise. Let’s say you have a MIDI track and an audio track with a UAD-1 plug-in on the audio track insert.
Important: UAD TrackAdv should not be used in host applications that provide full PDC throughout the entire signal path. TrackAdv or DelayComp is not needed at all in such hosts. See “Host PDC Implementation” on page 101. How to use TrackAdv UAD TrackAdv is designed to be used on audio track inserts of tracks that are assigned to a send/group/bus that has one or more UAD plug-ins applied.
Send Situation: You have a song with drums and guitar on audio tracks, and a MIDI bass line. You want a cohesive room reverb on the audio tracks so you send them to the UAD RealVerb Pro via an effect send. Result: The RealVerb Pro effect return plays late in relation to the MIDI track. Solution: Put a TrackAdv plug-in on the track insert of the audio tracks with a Plugs value of 1. If you had an 1176LN and a RealVerb Pro on the send return, the TrackAdv Plugs value would be 2.
• Compensating for upsampled UAD plug-ins is not required when these plugins are used for program material on the output bus, where latency is not a consideration. • If only one latency value is shown in Table 7 on page 110, the value applies to both UAD–1 and UAD-2. N/A means that sample rate is not supported. • UAD Pultec-Pro only requires one compensation per instance. For example, if using both MEQ-5 and EQP-1A within a single Pultec-Pro, only one instance compensation is required.
Upsampling Values Table Table 7 below lists the additional latency produced by upsampled UAD plugins. See the “Compensating Upsampled Plug-Ins” on page 108 for details. Table 7. Upsampled UAD plug-ins with additional latencies Sample Rate (kHz) Upsampled Plug-In Pultiec EQ/Pultec-Pro 44.1 UAD-1: 13 UAD-2: 31 48 UAD-1: 13 UAD-2: 31 88.2 13 96 13 176.4 0 192 0 Precision Limiter 64 69 129 140 259 281 Neve 33609 FATSO Jr./Sr.
Upsampled Compensation Examples Group/Bus Example Situation: You have a session at 44.1kHz with bass, drums, guitar, and 2 vocal tracks. You want a fat, warm vocal blend so you put both vocal tracks on a group/bus and apply one instance of UAD Helios 69 to the vocal bus. Result: The vocal tracks play late in relation to the instrument tracks.
PMB Group/Bus Examples Situation: You have a session with bass, drums, piano, and 2 vocal tracks. The session is running at 44.1kHz and your I/O buffer is set to 512 samples. You want to tighten up the rhythm section so you put the bass, drum, and piano tracks on a group/bus and apply one instance of Precision Multiband to the rhythm section group/bus. Result: The rhythm section plays late in relation to the vocal tracks.
Live Processing The previous discussions of delay compensation applies primarily to playback and mixing of existing tracks. During recording (tracking), the primary concern usually centers around getting the absolute lowest possible latency out of your hardware and software combination. The lower the latency is, the closer you can get to a realtime, “ears match the fingers” performance situation in the digital environment where some latency is unavoidable.
CHAPTER 10 UAD ExpressCard Products Overview ExpressCard™ is a computer expansion slot typically used in notebook computers. Utilizing a 2.5Gbps differential serial link, ExpressCard conveniently leaves slower USB and Firewire connections free for dongles, Audio I/O, and hard drives, while operating at full PCI Express bandwidth. Universal Audio has three ExpressCard products: UAD-2 SOLO/Laptop™, UAD-Xpander™, and UAD–Xtenda™.
Instructions in this chapter are specific to the UAD ExpressCard products only, and apply in addition to the other chapters. To get the most from your UAD ExpressCard products, please review the following additional information: Important Information Important user information is presented on a printed document within the UAD ExpressCard product retail package. It provides details that may not be included in this manual. Please review the information carefully.
UAD-2 SOLO/Laptop Details Important: The information in these sections apply to UAD-2 SOLO/Laptop ONLY. For UAD–Xpander instructions, see “UAD–Xpander Details” on page 121. Features • UAD-2 Solo in a self-contained, compact ExpressCard/34 package (can be used in ExpressCard/54 slots with included adapter) • 44.1-192k DSP Audio Accelerator for notebook computers • Ultra-fast (2.
Software Updates The CD-ROM in the retail package may not contain the latest UAD software. Please check our website for software updates: • http://www.uaudio.com/support/uad/downloads/ • If the UAD software is already installed, the UAD Meter & Control Panel has a convenient button that links to the updates page (“Check for Updates” on page 65). ExpressCard/34 to ExpressCard/54 Adapter The UAD-2 SOLO/Laptop device uses the ExpressCard/34 form factor.
3. Slide the UAD-2 SOLO/Laptop ExpressCard fully into the host computer’s ExpressCard/34 expansion slot (or ExpressCard/54 expansion slot with the adapter) until it is firmly seated. 4. The LED on the SOLO/Laptop will initially flash red/green, indicating the hardware is powered by the computer but the UAD drivers are not yet loaded.
Important SOLO/Laptop Notes For maximum system stability and reduced possibility of data loss, please observe these operational requirements when operating UAD-2 SOLO/Laptop: Sleep Important: Close sessions and remove UAD-2 SOLO/Laptop before sleep! Before putting the computer into system sleep/hibernate/standby mode, quit all UAD host software and remove the UAD-2 SOLO/Laptop device. Otherwise, unpredictable behavior and/or loss of session data could occur.
UAD–Xtenda UAD–Xtenda is an optional package that enables UAD-2 SOLO/Laptop and UAD–Xpander to be used in a desktop computer that has PCIe expansion slots. The UAD–Xtenda provides a convenient way to switch the same UAD ExpressCard between desktop and notebook computers. Note: UAD-Xtenda is a discontinued product.
UAD–Xpander Details Important: The information in the remainder of this chapter apply to the discontinued UAD–Xpander ONLY. For UAD-2 SOLO/Laptop instructions, see “UAD-2 SOLO/Laptop Details” on page 116. Features • Noiseless, Fanless, Alumi-cool chassis design • Ultra-fast (2.5Gbps) ExpressCard/34 interface (can be used in ExpressCard/54 slots) • 44.
UAD–Xpander requires the following hardware and software: • UAD Powered Plug-ins software v4.7.1 or higher (included) • Available ExpressCard/34 or ExpressCard/54 expansion slot • Available PCI Express (PCIe) slot for each optional UAD–Xtenda card • (Win) Windows notebooks are supported under Windows Vista only Software Installation Install Software First Software installation for UAD ExpressCard products is the exact same procedure as those for UAD cards. Install the software before the hardware.
UAD–Xpander Connections This section describes how to set up the UAD–Xpander and connect it to the computer. Hardware installation is the same for all platforms. Power Down! Before connecting the UAD–Xpander to your computer, make sure both systems are completely powered down. Important: Because the Xpander drivers are loaded during startup, merely putting the system to hibernate or sleep is insufficient. To connect the UAD–Xpander components: 1.
Note: All UAD–Xpander instructions apply regardless of whether the unit is attached to a notebook computer or a desktop computer via UAD–Xtenda. Startup Sequence For optimum results, specific steps must be followed when starting the UAD–Xpander and host computer system. Following this sequence ensures the Xpander electrical and UAD driver requirements are properly met. UAD–Xpander Pre-Flight Check 1. Ensure UAD Powered Plug-Ins v4.7.
• When UAD–Xpander power is on but there is no electrical connection to the computer (such as when the computer is off), the UA logo glows red. • When power is on and there is an electrical connection to the computer, the UA logo glows blue (Figure 25). Figure 25. UAD–Xpander Power Indicator Note: The Power Indicator displays the state of the electrical connections only. It does not indicate when UAD driver communication is established (use the UAD Meter to check driver communication status).
2. Power down the ExpressCard, using the following method as defined by the operating system: • (Windows) In the Windows Task Bar, click the Safely Remove Hardware icon, and select “Safely remove Universal Audio UAD-1 DSP card” (Figure 26 on page 126). • (Mac) In the Menu Bar, click the ExpressCard status menu icon, and select “Power off Card” (Figure 27 on page 126) 3. Turn off the power switch on the back of the UAD-Xpander. 4. Remove the UAD ExpressCard from the expansion slot.
To disable Sleep in Windows: 1. In the Control Panel>Performance and Maintenance>Power Options>Power Scheme panel, set the Power Scheme to “Always On” 2. To disable Sleep in when notebook lid is closed: In Control Panel>Performance and Maintenance>Power Options: Click the “Change when the computer sleeps” option on the left side of the screen, then select “Never” Mac System Sleep is supported under Mac OS X.
CHAPTER 11 UAD-2 Satellite Overview UAD Powered Plug-Ins via FireWire UAD-2 Satellite is a UAD-2 DSP accelerator that connects to the host computer via FireWire 800 or FireWire 400 and hosts UAD-2 Powered Plug-Ins on Mac OS systems. Unlike previous UAD-2 PCIe products, UAD-2 Satellite is a completely external unit with power supply that does not require a PCI, PCIe or ExpressCard interface.
Technical Updates Updated technical information and the latest UAD software is regularly posted to the UA website. Please visit the UAD support pages for the latest UAD-2 Satellite support bulletins: • http://www.uaudio.com/support/uad/satellite-support UAD-2 Satellite Instructions All instructions that apply specifically to UAD-2 Satellite are in the “UAD-2 Satellite Installation” section beginning on page 130.
Satellite System Requirements To use UAD-2 Satellite, your system must meet the minimum system requirements for UAD Powered Plug-Ins in addition to the UAD-2 Satellite product-specific requirements below. UAD-2 Satellite requires the following hardware and software: • Mac OS 10.6 Snow Leopard (Intel systems only) • UAD Powered Plug-ins software v5.8.
Security Slot The Kensington Security Slot is on the rear panel. This feature helps prevent theft when used with an optional locking cable such as the Kensington MicroSaver Security Cable. Refer to the instructions that come with the optional cable. External Power Supply UAD-2 Satellite includes an external international DC power supply with changeable AC connectors to match the AC socket in various countries (Figure 28). Figure 28.
Connecting UAD-2 Satellite Important: Install the UAD software before connecting UAD-2 Satellite (see “UAD Software Installation” on page 28). UAD-2 Satellite may be connected before or after the system is booted. To connect UAD-2 Satellite: 1. Ensure that UAD v5.8.1 (or higher) software is already installed and that the system was restarted after software installation. See “UAD Software Installation” on page 28 for specific instructions. 2. Ensure that all UAD plug-in host software is quit. 3.
UAD-2 Satellite Operation After UAD Powered Plug-Ins software v5.8.1 (or higher) is installed (page 28) and UAD-2 Satellite is properly connected to the computer via FireWire (page 130), the system is ready for use. Refer to other chapters in this manual for software operating instructions. All UAD-2 Satellite operations (except for hardware setup) are similar to the operation of other UAD-1/UAD-1e/UAD-2 devices. See “Important UAD-2 Satellite Notes” on page 135 for exceptions specific to UAD-2 Satellite.
Existing UAD Licenses If you already have another UAD device, you can share those UAD licenses with UAD-2 Satellite. This option is presented during initial device registration; please consider your options carefully when registering. Hot Plugging Hot plugging refers to the ability to disconnect UAD-2 Satellite while the host computer is active or sleeping, and reconnect it at a later time without rebooting. UAD-2 supports hot plugging (hot plugging is part of the FireWire specification).
System Sleep System sleep while UAD-2 Satellite is active is not supported. Before sleeping the system, quit all plug-in host applications (including the UAD Meter & Control Panel) and disconnect UAD-2 Satellite. After wake, UAD-2 Satellite may be reconnected. Important: Quit all hosts and disconnect UAD-2 Satellite before system sleep.
• Although “Hot Plugging” on page 134 is supported, disconnecting UAD-2 Satellite when UAD plug-ins are running could cause unpredictable behavior. Quit all UAD hosts and the UAD Meter & Control Panel before disconnecting or powering down UAD-2 Satellite. Important: Do not disconnect UAD-2 Satellite while UAD plug-ins are loaded. • Once the host software is quit, you may connect or disconnect the unit even if the host computer and/or UAD-2 Satellite is powered on (see “Hot Plugging” on page 134).
FireWire Basics FireWire (also known as “IEEE 1394” and “i.Link”) is a high-speed serial data interconnection protocol that is used to transfer digital information between devices. FireWire is commonly used to interconnect computer systems to hard drives, audio interfaces, and digital camcorders. A complete discussion of FireWire is beyond the scope of this manual, but some of the main points and how they apply to UAD-2 Satellite are covered below. FireWire vs.
Note: UAD-2 Satellite is a FireWire 800 device. See “Mixing FireWire Speeds” on page 140 for more information about using FW800 and FW400 devices on the same FireWire bus. FireWire Connectors FireWire 400 and FireWire 800 devices use different connectors (Figure 32 below). The connectors are not interchangeable; this helps to differentiate between the two device speeds. FW 800 (9-pin) FW 400 (6-pin) FW 400 (4-pin) Figure 32.
Additionally, UAD-2 Satellite can function as a FireWire hub, by using the unused ports on the unit to connect other FireWire devices as in Figure 36. Note that UAD-2 Satellite does not supply FireWire bus power to downstream devices; see “FireWire Bus Power” on page 144 for details. Computer FW Hub FW Audio Interface External HD UAD-2 Satellite Figure 33. FireWire bus connections via a hub Computer UAD-2 Satellite FW Audio Interface External HD Figure 34.
Connections The ability to connect and disconnect FireWire devices while power is applied (or not) is part of the FireWire specification. Since UAD-2 Satellite is IEEE 1394 compliant, the device may be “hot plugged” while powered up. Note: See “Hot Plugging” on page 134 for detailed information.
It is possible to configure a FireWire bus to run at both FW400 and FW800 speeds simultaneously if the host computer bus is FW800, supporting maximum throughput for a mix of FW400+FW800 devices. This is accomplished by putting any/all FW400 devices AFTER any/all FW800 devices in a daisy chain (see Figure 39 on page 141).
Figure 40. The UAD Meter with UAD-2 Satellite Important: FireWire bandwidth is unrelated to UAD DSP loads. See “FireWire Bandwidth vs. UAD DSP” on page 143 for details. Sharing Bandwidth If UAD-2 Satellite is the only device on the FireWire bus, UAD processing traffic can use the entire bandwidth of the bus. However, if other FireWire devices are on the bus, the amount of bandwidth used for UAD traffic must be reduced so those devices will have enough bandwidth for their processes as well.
Figure 42. The UAD FireWire panel (control descriptions are in Chapter 6) FireWire Bandwidth vs. UAD DSP The amount of FireWire bandwidth used by UAD-2 Satellite depends on the number (the quantity) of UAD plug-ins that are loaded on the device; the amount of DSP used by a UAD plug-in (the quality) does not affect FireWire bandwidth at all. In other words, each UAD-2 Satellite plug-in instance uses a fixed amount of FireWire bandwidth, regardless of how much DSP a given UAD plug-in uses.
FireWire vs. PCIe The maximum possible data throughput even at FireWire 800 speeds is only a fraction of what is possible with PCIe. Therefore, if very high UAD plug-in counts are required, UAD-2 PCIe cards provide an alternate solution. FireWire Bus Power Some FireWire devices can be “bus powered” which means the device derives its operating electricity from the FireWire bus itself without a power supply of its own.
FW800 Computer The externally powered FireWire repeater must supply bus power! Powered FW400 repeater UAD-2 Satellite (externally powered) External HD 1 (externally powered) External HD 2 (bus powered) Apogee Duet (bus powered) Figure 43. Proper connection of bus powered FireWire devices This example uses the Apogee Duet, a popular audio interface with one FireWire port that can only be bus powered.
CHAPTER 12 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 44. The UAD Ampex ATR-102 plug-in window Figure 45.
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 46. 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 157) 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 (page 74) is disabled. If DSP LoadLock is enabled (the default setting), activating OFF will not reduce DSP usage.
Each type has its own subtle sonic variation, distortion onset, and tape compression characteristics. Generally speaking, the lower the Cal Level for each formula, the higher the signal level required to reach saturation and distortion. 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.
The tape manufacturer’s recommended calibration settings for each Tape Type are shown in Table 8. Table 8.
Repro Repro mode models the sound of tape recording through the record head and playback through the reproduction head, plus all corresponding machine electronics. Input Input mode emulates the sound of the Ampex ATR-102 through the machine electronics only, without tape sonics. This is the scenario when the machine is in live monitoring mode but the tape transport is not running. Thru Thru is a processor bypass control.
Secondary Controls The secondary controls (Figure 47 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 47.
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 13 below. Table 13.
Original Ampex ATR-102 Mastering Recorder Brochure UAD Powered Plug-Ins Manual - 172 - Chapter 12: Ampex ATR-102
CHAPTER 13 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 49. 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 50.
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 14. 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.
Parametric Q The Q (bandwidth) knob sets the proportion of frequencies surrounding the center frequency to be affected by the gain control. The Q range is 0.25–16; higher values yield sharper slopes. 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.
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 54. Figure 54. 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 57 on page 183) when shelf mode is active.
Figure 56. Shelf Type A Shelf Mode Indicator Line Figure 57. Shelf Type B Figure 58.
CHAPTER 14 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 189) 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 61. 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 186). 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 62. The original Cooper Time Cube hardware front panel Figure 63.
CHAPTER 15 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 65. 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 66. 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, “Tempo Sync”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 67. The DM-1L includes a Link button RS-1 Reflection Engine Figure 68. 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, “Tempo Sync” for more information. Shape Pop-up Menu Determines the shape of the reverberant space, and the resulting reflective patterns. Table 15.
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 16 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.
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. Output does not affect the amount of compression. The Meter buttons define the mode of the VU Meter. The buttons do not change the sound of the signal processor. The active button has a darker appearance when compared to the inactive buttons.
CHAPTER 17 DreamVerb Overview DreamVerb™, Universal Audio’s flagship stereo reverb plug-in, draws on the unparalleled flexibility of RealVerb Pro. Its intuitive and powerful interface lets you create a room from a huge list of different materials and room shapes. These acoustic spaces can be customized further by blending the different room shapes and surfaces with one another, while the density of the air can be changed to simulate different ambient situations.
Screenshot Figure 70. The DreamVerb plug-in window Signal Flow Figure 71 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 71.
The DreamVerb user interface (Figure 70 on page 204) 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 74 on page 207). Early reflection pre-delay, slope, timing, and amplitude are specified in the Reflections panel (Figure 76 on page 212).
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 72. 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 73.
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 74 on page 207) and Reverberation panel (Figure 77 on page 213) 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 75 on page 209) 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 76. 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 77) 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 78. 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 216) 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.
Windows On Windows systems, the default preset location is inside the Presets directory, which is created within the directory selected during software installation. For example, if the default location was specified when running the UAD Powered Plug-Ins Installer, the location would be: C:\Program Files\Universal Audio\Powered Plugins\Presets This default location can be changed during installation; the new location will be remembered as the default.
Space In some sense, Shape determines the spatial characteristics of the reverberator, whereas Materials effects the spectral characteristics. Preset Design Tips Here are some practical tips for creating useful reverbs with DreamVerb. These are not rules of course, but techniques that can be helpful in designing the perfect sonic environment.
• 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. Higher diffusion values yield a fuller sound, good for percussive sounds; lower diffusion values yield a less dense sound, good for vocals, synths, etcetera. • When monitoring your preset, try switching from Dry solo, Wet solo, and a useful mix. Solo the reflections and reverberation, and disable/enable EQ.
CHAPTER 18 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 80. The FATSO Jr. plug-in window Figure 81. 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 227).
Table 16. 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 223 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 226). 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 223 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 226), and the sensitivity of the Warmth processor (page 227). 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 228).
CHAPTER 19 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” on page 92.
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” on page 74 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 20 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 83. 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 “Controls View” on page 87), 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 242 and Table 17 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 17 displays the parameter that each control is mapped to for each of the EMT 250 program modes. See “Variable Control Functions” on page 242 for details. Table 17.
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 243 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 243 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 243 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 243 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.
Original EMT 250 advertisement – “A dream becomes reality” The EMT 250 Electronic Reverberator hardware unit Special thanks to EMT Studiotechnik GmbH, Dr. Barry Blesser, Allen Sides, and Ocean Way Recording for their assistance with this project.
CHAPTER 21 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 84. 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 257). The parameter can be adjusted by using the metallic “slider handle” or the “slider nose” (both sliders control the same parameter; see Figure 85). Figure 85.
The available delay range is 80 to 700 milliseconds. When Sync is active, beat values from 1/64 to 1/2 can be selected (see Table 5 on page 97). 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, “Tempo Sync” for detailed 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, “Tempo Sync” for detailed 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 22 Fairchild 670 Overview 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 compressor/limiters, which are famous for their fantastic sound quality.
Fairchild Screenshot Figure 86. The Fairchild plug-in window 2 Compressors, 4 Modes There are two compressors within the Fairchild 670. They can be used as dual L/R, dual mono/stereo, or they can be linked together and used on either the L/R or mono/stereo signals. The mode in which the compressors operate is determined by the combination of the AGC switch and the Sidechain Link switch. See “Fairchild Modes” on page 265 for detailed mode information.
Controls Overview Most of the controls are associated with one or the other of the compressors, as opposed to being strictly associated with one channel of input/output (depends on active mode). These controls include Threshold, Time Constant, Bias Current Balance, and DC Bias. There are two sets of controls that always work on the left and right signals: input level and output level. In Lat/Vert mode, left is the mono input, and right is the stereo input.
Fairchild Modes Dual Left/Right In Dual L/R mode, the Fairchild operates as two monophonic compressors with completely independent controls for the left and right channels. There is no interaction between the left and right channels. Lateral – Vertical In Lat/Vert mode, the 670 acts on the lateral and vertical (the sum and difference) components of the two stereo channels.
Stereo, coupled mono/stereo This mode, like stereo couple left/right, causes the two compressors to be linked together so that they always compress the same amount. But here, the inputs to the two compressors are fed with the mono and stereo components of the signal. This means that in general a transient which occurs in both channels will cause a bit more compression than a transient which only appears on left or right.
Meter Select Switch This switch determines what is displayed on the VU meters. If GR is selected, the meter will show gain reduction in dB for the corresponding compressor channel (which is not necessarily left or right; depends on the active mode). If the AGC switch has been set to left/right, the GR shown will be for the left or right channel. If the AGC switch has been set to lat/vert, the GR shown will be for the mono or stereo channel. In GR mode, the upper labels show gain reduction in dB.
Time Constant This 6-position switch provides fixed and variable time constants (attack and release times) to accommodate various types of program material. Positions 1-4 provide successively slower behavior, and 5 and 6 provide program dependent response. The values published by Fairchild for each position are in Table 19 below. The actual measured times are a bit different, but the overall trend is the same. Table 19.
DC Bias DC Bias controls the ratio of compression as well as the knee width. As the knob is turned clockwise, the ratio gets lower and the knee gets broader. The threshold also gets lower as the knob is turned clockwise. The ‘factory cal’ tick mark should be aligned with the screw slot “dot” for factory specification. It would probably be 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.
CHAPTER 23 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. This behavior is identical to the original hardware, which we modeled exactly. When the plug-in is viewed in parameter list mode (see “Controls View” on page 87), the actual parameter values are displayed. The Power switch determines whether the plug-in is active. Click the button to toggle the state.
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 88. 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 “Compensating Upsampled Plug-Ins” on page 108 for more information. The Harrison 32C SE does not require additional latency compensation because it is not upsampled.
CHAPTER 24 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” on page 74 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 “Compensating Upsampled Plug-Ins” on page 108 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 - 280 - Chapter 24: Helios Type 69 Equalizer
CHAPTER 25 LA-2A and 1176LN Overview The LA-2A and 1176LN compressor/limiters long ago achieved classic status. They're a given in almost any studio in the world – relied upon daily by engineers whose styles range from rock to rap, classical to country and everything in between. With so many newer products on the market to choose from, it's worth looking at the reasons why these classics remain a necessary part of any professional studio's outboard equipment collection.
Figure 91 depicts the input and output characteristics of a compressor and perfect amplifier. When operated within its specified range, an amplifier provides a constant amount of gain regardless of the input signal level. In Figure 91, the signal level of a perfect amplifier is represented with a constant output gain of 10 dB. In this example, a signal with an input level of –30 dB results in an output level of –20 dB, which is an increase of 10 dB.
Compression region 10 dB of compression +10 2:1 compression ratio 0 Output Level (dB) –10 knee –20 –30 –30 –20 –10 0 +10 Input Level (dB) Figure 92. Input and output curve of compressor with 2:1 ratio and –20 dB threshold As mentioned previously, the compression ratio is defined as the ratio of the increase of the level of the input signal to the increase in the level of the output signal. In Figure 92, the input level is increased by 10 dB while the output level increases 5 dB.
Teletronix LA-2A Leveling Amplifier Background Audio professionals passionate about their compressors revere the LA-2A. The original was immediately acknowledged for its natural compression characteristics. A unique electro-optical attenuator system allows instantaneous gain reduction with no increase in harmonic distortion – an accomplishment at the time, still appreciated today.
LA-2A Controls Figure 94. The LA-2A plug-in window. Limit/Compress Changes the characteristics of the compressor I/O curve. When set to Compress, the curve is more gentle, and presents a low compression ratio. When set to Limit, a higher compression ratio is used. Gain Adjusts the output level (by up to 40 dB). Make sure to adjust the Gain control after the desired amount of compression is achieved with the Peak Reduction control. The Gain control does not affect the amount of compression.
1176LN Solid-State Limiting Amplifier The 1176LN is known for bringing out the presence and color of audio signals, adding brightness and clarity to vocals, and “bite” to drums and guitar. 1176LN Signal Flow A functional block diagram of the 1176LN Limiting Amplifier is provided in Figure 95. Signal limiting and compression is performed by the Gain Reduction section. Before the signal is applied to the Gain Reduction section, the audio signal is attenuated by the Input stage.
1176LN Controls Figure 96. The 1176LN plug-in window Input Adjusts the amount of gain reduction as well as the relative threshold. An Input value of ∞ (turned fully counterclockwise) yields no compression (and no signal level). Rotate this control clockwise to increase the amount of compression. Output Adjusts the output level (by up to 45 dB). Make sure to adjust the Output control after the desired amount of compression is achieved with the Input and Attack controls.
All Buttons mode Just like the hardware version of the 1176LN, it is possible to depress all the Ratio buttons simultaneously, a well-known studio trick. In this mode, the ratio is around 12:1, and the release happens faster, and the shape of the release curve changes. With lower amounts of compression, the attack is delayed slightly, as there is a slight lag before the attack attenuated the signal. That attack value remains at whatever the value is on the Attack control.
Stereo Operation Phase-coherent stereo imaging is maintained when the 1176LN plug-in is used on a stereo signal. 1176SE “Special Edition” Figure 97. The 1176SE plug-in window Overview The 1176SE is derived from the 1176LN. Its algorithm has been revised in order to provide sonic characteristics similar to the 1176LN but with significantly less DSP usage. It is provided to allow “1176LN-like sound” when DSP resources are limited. The 1176SE behavior is practically identical to the 1176LN.
CHAPTER 26- 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 “Compressor Basics” on page 281. Comp/Lim This switch changes the characteristics of the compressor I/O curve. When set to Compress, the curve is more gentle, and presents a low compression ratio. When set to Limit, a higher compression ratio is used. Gain The Gain knob adjusts the output level (by up to 50 dB).
CHAPTER 27 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 99.
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 21 on page 311 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 101) are used to specify which of the nine default Lexicon programs, and its associated algorithm, is active. See “Lexicon 224 Programs” on page 295 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 306 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 304 for parameter details.
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. Note: When link is active, modifying the right channel parameters will have no effect when changed from a control surface or when in “controls only” (non-GUI) mode. Bug Fixes The original Lexicon 224 code contains programming errors in the Hall B and Chorus algorithms.
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 21. 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 22 below lists the default values of all available parameters for each program. Table 22.
CHAPTER 28 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 103 on page 312 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 29 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 104. 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.
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 four available center frequencies, and resulting frequency ranges, are shown in Table 23 below. A switch is “ON” when its LED is red. A green LED indicates the switch is “OFF.” Table 23.
CHAPTER 30 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 105. The Massive Passive plug-in window Figure 106. 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 329 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 24 below. Table 24. 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 324) 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 108 below. Figure 107. Effect of Bandwidth control on response curve in Bell mode Figure 108.
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 25 below. Frequency Available Frequencies Table 25.
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 109 below.
Figure 109. 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.manleylabs.com/techpage/manuals.
CHAPTER 31 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 110. 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 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. When Mix is at zero, the Drive/Gain control (and Boost on non-SE version) are still active and audible. Setting Mix at zero is the same as setting the Effect/Bypass switch to Bypass.
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 “Compensating Upsampled Plug-Ins” on page 108 for more information. The Moog Filter SE does not require additional latency compensation because it is not upsampled.
CHAPTER 32 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 112. 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” 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. When Sync is active, the LFO speed is shown as a beat division (or multiplier).
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 “Compensating Upsampled Plug-Ins” on page 108 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. See “Host PDC Implementation” on page 101 for more information.
CHAPTER 33 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” on page 74 is enabled). Note: 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 1073SE Controls The Neve 1073SE controls are exactly the same as the Neve 1073. Please refer to the Neve 1073 section for Neve 1073SE control descriptions (see “Neve 1073 and 1073SE Controls” on page 349). 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.
CHAPTER 34 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 116 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 116 on page 355). 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 117. 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 35 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. (UAD-2 only) UAD-2 DSP usage is reduced only when DSP LoadLock (page 74) 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 “Compensating Upsampled Plug-Ins” on page 108 for more information. The Neve 31102SE does not require additional latency compensation because it is not upsampled.
CHAPTER 36 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 120. 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 372). 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.
Technical Article The article “Ask the Doctors: Modeling of the Neve 33609 compressor/limiter” contains interesting technical details about the 33609. It is available at our online webzine: • http://www.uaudio.com/webzine/2006/august/index2.html Neve 33609 and 33609SE Controls Limiter Controls in this section only function when the limiter is enabled with the “limit In” switch (the Power switch must also be on). The compressor precedes the limiter (see “Signal Flow” on page 368).
Compressor Controls in this section only function when the compressor is enabled with the “compress In” switch (the Power switch must also be on). Note: The compressor precedes the limiter (see “Signal Flow” on page 368). Compressor Threshold Threshold determines how much compression will occur. When the input signal exceeds the threshold level, the compressor engages. A smaller value results in more compression. The available range is from –20 dB to +10 dB, in 2 dB increments.
Other Controls The interface elements that are not directly contained within the compressor or limiter are detailed below. Output Gain This control is a software-only addition not found on the original hardware. It is an overall makeup gain stage at the output of the plug-in to compensate for reduced levels as a result of compression and/or limiting. The available range is –2 to +20 in 1 dB increments.
Link When set to link (down position), modifying any channel one or channel two control causes its adjacent stereo counterpart control to snap to the same 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.
• If you feed the same signal into both channels, you can have a lower threshold with a lower ratio on one channel, and a higher threshold with a higher ratio on the other channel. In this case, you will get a double knee, with the lower ratio being used between the knees, and the higher ratio above both knees. • If you feed the same signal into both channels, you can have a lower threshold with a faster release on one channel, and a higher threshold with a slower release in the other channel.
Headroom Switch The Headroom switch is provided to accommodate applications where high amounts of gain reduction are not desired. Headroom simply lowers the internal operating level so that the plug-in is not “pushed” into gain reduction as much. Headroom can be set to 22 dB, 18 dB, or 14 dB. At 22 dB, signals will push the plug-in into gain reduction (and more non-linearity and “good” harmonic distortion) more easily. Set the switch to a lower value when less gain reduction and color is desired.
Keep in mind there are no hard and fast rules. Use the above recommendations as guidelines and feel free to experiment with the various positions of the headroom switch regardless of the audio source. If it sounds good, use it! Note: Factory Presets The UAD Neve 33609/33609SE includes a bank of factory presets. These presets can be useful starting points for your particular source audio. The factory preset names begin with MSTR, BUSS, or TRAK. These indicate the setting of the headroom parameter.
Neve 33609SE Figure 122. The Neve 33609SE plug-in window Overview The UAD Neve 33609SE is derived from the UAD Neve 33609. Its algorithm has been revised in order to provide sonic characteristics very similar to the 33609 but with significantly less DSP usage. It is provided to allow 33609like sound when DSP resources are limited. Nobody with “golden ears” will claim it sounds exactly like the 33609, but it still sounds great and is very usable in most situations.
CHAPTER 37 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 123.
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 125 on page 380), and hysteresis is available (see Figure 126 on page 381). 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 126 on page 381.
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 382). 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. Note: For additional information, see “Compressor Basics” on page 281.
As the Threshold control is increased and more processing occurs, output level is typically reduced. Adjust the Gain control to modify the output of the module to compensate if desired. Note: The –20 dB switch increases the sensitivity of the limiter/compressor by lowering the range of the available threshold values. When –20 dB mode is active, the threshold range is 0 dB to –30 dB. When –20 is inactive, the threshold range is +20 dB to –10 dB.
Slower release times can smooth the transition that occurs when the signal dips below the threshold, which is especially useful for material with frequent peaks. However, if the release is too long, compression for sections of audio with loud signals may extend to sections of audio with lower signals. Fast release times are typically only suitable for certain types of percussion and other instruments with very fast decays. Using fast settings on other sources may produce undesirable results.
88RS EQ Band Layout EQ module enable switch High Frequency (HF) band controls High Midrange Frequency (HMF) band controls Low Midrange Frequency (LMF) band controls Low Frequency (LF) band controls Figure 127. Neve 88RS EQ Controls Layout This button activates the equalizer module. The module is active when the button is gray and the green indicator illuminates. EQ Enable (EQ) You can use this button to compare the equalized signal to the original signal or bypass the EQ altogether.
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 filter slope of the HF band can be changed with this control. When Hi-Q is off, the Q is 0.7. When Hi-Q is active, the Q is 2. Higher Q values mean the peak (or trough) has steeper slopes.
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. LMF Gain This control determines the amount by which the frequency setting for the LMF band is boosted or attenuated. The available range is ±20 dB. LMF Q The Q (bandwidth) control defines the proportion of frequencies surrounding the LMF band center frequency to be affected by the band gain control.
This button activates the cut filter. The cut filter is active when the button is gray and the red indicator illuminates. Cut Enable This knob determines the cutoff frequency for the cut filter. The available range is 7.5 kHz to 18 kHz for the high cut filter (lighter blue control), and 31.5 Hz to 315 Hz for the low cut filter (darker blue control). Cut Frequency Global This control enables the UAD Neve 88RS sidechain function.
The Power switch determines whether the plug-in is active. This is useful for comparing the processed settings to the original signal or bypassing the plug-in to reduce the UAD DSP load (load is not reduced if “UAD-2 DSP LoadLock” on page 74 is enabled). Power 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.
CHAPTER 38 Nigel Introduction Nigel offers the latest generation of guitar processing technology integrated into a complete multi-effects plug-in solution. Utilizing Universal Audio’s exclusive component modeling technology, along with some very creative digital design, Nigel delivers a complete palette of guitar tones along with most every effect a guitar player might need, all with minimal latency and no load on your host computer’s CPU.
Nigel Screenshot Figure 128. The Nigel plug-in window Nigel Modules Nigel is comprised of eight modules: Gate/Compressor, Phasor, Mod Filter, Preflex, Cabinet, Trem/Fade, Mod Delay, and Echo. In order to conserve UAD DSP resources when all of the modules are not required simultaneously, some of the Nigel components are also supplied as separate plug-ins.
Preflex Plug-in Preflex is the heart of Nigel. All of our plug-ins sound amazing but when it comes to guitar, Preflex really shines. This exciting new guitar processing technology offers truly dynamic sonic possibilities Multiple equalizers, amp types, and cabinets use sophisticated algorithms to provide analog sound quality never before available in a digital environment.
Gate/Comp Module Figure 130. The Gate/Comp module The Gate is the first sub-module in the Preflex signal chain. Its output is passed to the input of the Compressor. The compressor output is then passed to the input of the Amp module within Preflex. A gate stops the input signal from passing when the signal level drops below a specified threshold value.
Gate Fast Button The Fast control reduces the release time of the gate. It has no effect on the attack time. When enabled, the gate will release quickly. On signals that slowly decay and/or have a wide dynamic range, a smoother (less choppy) sound may be obtained with Fast mode turned off. Fast mode is engaged when the button indicator is bright red. The time values are 50ms when engaged and 170ms when off. Gate Threshold Knob Sets the threshold level for the gate.
Compressor Attack Menu Sets the amount of time that must elapse, once the input signal reaches the Threshold level, before compression will occur. The faster the Attack, the more rapidly compression is applied to signals above the Threshold. Three Attack values are available: Slow (50ms), Medium (8ms), and Fast (400µs). Compressor Release Menu Sets the amount of time it takes for compression to cease once the input signal drops below the Threshold level.
Amp Module The Preflex Amp is where Nigel’s real magic happens. Behind its deceptively simple user interface is “rocket science” in action. The input to the Amp module is received from the Compressor output. The Amp output is passed to the input of the Cabinet module. Figure 131. The Amp module within Preflex Amp Type and Variable Knob Functions The function of the amp knobs vary depending on the amp type. When an amp type is selected, Preflex is internally reconfigured.
Knob Values Are Offsets Knob settings do not change to new values when an amp type is selected. This is because knob values are not absolute. Instead, they are an offset to the factory programmed amp type value. For example, if Post-Lo EQ displays a value of 3.0, then 3 dB is added to the amp type internal (preset) value. Of course, knob settings do change when user settings are loaded.
Pre-EQ Mid Knob Modifies the middle frequency response of the signal before the Amp. The frequency that this knob controls is determined by the Color knob (see Color knob description for more details). Pre-EQ Hi Knob Modifies the high frequency response of the signal before the Amp. This knob behaves differently than the Lo and Mid knob. Rather than boosting or cutting the gain of a certain frequency, the Hi knob increases the amplifier's sensitivity to high frequencies.
Bright Button Increases the brightness of the Amp model. Bright is on when the button glows bright red. Amp On/Off Button Enables or disables the Amp module within Preflex. The Amp is engaged when the button indicator is bright red. You can use this switch to compare the Amp settings to the original signal or bypass the entire Amp section to reduce UAD DSP load.
Amp-A Type Menu Determines the amp type for the “A” section of the Amp. Selecting an Amp Type reconfigures the amplifier characteristics and the function of the other Amp parameters. Amp-B Type Menu Determines the amp type for the “B” section of the Amp. Selecting an Amp Type reconfigures the amplifier characteristics and the function of the other Amp parameters. Amp Morph Slider The Morph control is used to smoothly transform one amp type into another, creating new sounds never before possible.
For the following descriptions of the Cabinet models and other references that you may find throughout this manual, please be aware that Celestion, Greenback, Oxford Blue, Marshall, Fender, Line 6, Pod, SansAmp, Shure, ADA, Utah and any other manufacturer, model name, description, and designations are all trademarks of their respective owners, which are in no way associated or affiliated with Universal Audio.
Cabinet On/Off Button Enables or disables the Cabinet module within Preflex. The Cabinet is engaged when the button indicator is bright red. You can use this switch to compare the Cabinet settings to that of the original signal or bypass the entire Cabinet section to reduce UAD DSP load. Output Level Meter This LED-style VU meter displays the level of the signal at the output of the Cabinet. Just before the red ‘LED’ is illuminated, the signal is at 0 dB.
Sweep Lo Knob Sets the lowest frequency of the Phasor. The available range is from 50 Hz to 6000 Hz. Because the Sweep Lo frequency cannot be set higher than the Sweep Hi frequency, if the Lo value is increased beyond the Hi value the Hi value will increase to match the Lo value. Sweep Hi Knob Sets the highest frequency of the Phasor. The available range is from 50 Hz to 6000 Hz.
LFO Type Menu Determines the LFO (low frequency oscillator) waveshape and phase used to modulate the signal. The waveshape can be set to triangle or sine, each with varying duty cycles and phases. Table 29. Phasor LFO Types and Descriptions Sin Pure sine wave. Sin 2 Modified sine wave that stays high longer. Sin 3 Modified sine wave that stays low longer. Square Square wave. Square 2 Modified square wave that stays high longer. Square 3 Modified square wave that stays low longer.
Mod Filter Module The Mod Filter is an advanced filter plug-in that is capable of fixed-wah, autowah, envelope follower, sample/hold-driven filter, and other modulated filter effects. It has been modeled after the Mutron III and other popular filters. The filter cutoff frequency can be controlled by the signal level at the input to the module or a low frequency oscillator (LFO). This realtime dynamic response is what gives the Mod Filter its unique sound. Figure 134.
Sens/Rate/ Wah Knob The function and label of the first knob in the Mod Filter is determined by the Mod Type setting (see Figure 134 on page 407). When the Mod Type is an envelope, the label changes to “Sens” and determines the gain sensitivity of the Mod Filter. When the Mod Type is an LFO, the label changes to “Rate” and determines the rate of the LFO. When the Mod Type is set to Wah, the label changes to “Wah” and adjusts the wah pedal position.
Sweep Hi Knob Sets the highest frequency to be affected by the Mod Filter. The available range is from 50 Hz to 4000 Hz. Because the Sweep Hi frequency cannot be set lower than the Sweep Lo frequency, if the Hi value is decreased below the Lo value the Lo value will decrease to match the Hi value. Resonance (Res) Knob Sets the amount of filter intensity for the Mod Filter. A higher value will deliver a sharper, more pronounced effect. Output Knob Adjusts the signal output level of the Mod Filter.
Wah Pedal Mode Similar to Wah mode, in Wah Pedal mode the filter cutoff frequency is varied according to the Wah knob setting. However, when the knob reaches its maximum value the effect is bypassed until the knob reaches is maximum value again at which time the effect is re-engaged. Wah Pedal mode is ideally suited to emulating a real Wah pedal by using a MIDI foot pedal controller. Mod Menu Table Table 30. Mod Filter: Mod Types and Descriptions Filter Type Menu Sin LFO mode with Sine waveshape.
TremModEcho plug-in The TremModEcho is loaded as one plug-in but consists of three modules: Trem/Fade, Mod Delay, and Echo (Figure 135). Each of the module controls is described in the following pages. Figure 135.
Trem/Fade Module Figure 136. The Trem/Fade module Trem/Fade is a sophisticated envelope-controlled modulation processor that can produce classic tremolo, fade, and other gain modulation effects. Tremolo is achieved by modulating the amplitude (volume) of a signal with a low frequency oscillator (LFO). Trem/Fade includes some new modes such as Shimmer and VariTrem that enable the production of new volume effects. Sync Button This button puts the plug-in into Tempo Sync mode.
Fade In Knob Determines the signal fade in time. Fade In is typically used to create automatic volume swells. The range is from None to 4000 milliseconds. When set to None, there is no fade in and only the Tremolo effect is active. Onset Knob Determines the time for the Tremolo effect to reach the specified depth. Onset behaves as an intensity ramp for the Tremolo effect. The range is from None to 4000 milliseconds.
Tremolo Mode When Tremolo mode is selected, the Fade In and Onset controls are set to zero and the Trem/Fade module behaves as a ‘normal’ tremolo effect. However, the Fade In and Onset controls are still active and can be adjusted as desired. Two Tremolo modes are available. Each has different settings but the controls behave exactly the same in both modes. If the Depth value is zero and/or the Threshold value is set too high in Tremolo mode, you will not hear the tremolo effect.
Mod Delay Module Figure 137. The Mod Delay module The label and function of the second two knobs depend upon the Mode menu selection. The Mod Delay is a short digital delay line that includes a low frequency oscillator. The Mod Delay produces lush chorus, flange, and vibrato effects. Because the Trem/Fade amplitude processor can be used to control the Mod Delay, sophisticated envelope-controlled flange, chorus, and vibrato modulations can be achieved.
If the LFO Type menu is set to one of the Trem modes, the Rate is linked to the Trem/Fade module rate. In this scenario the Rate knob value changes to “Trem”, adjusting the Mod Delay Rate will have no effect, and the modulation rate is determined by the Trem/Fade module settings (even if the Trem/Fade module is disabled with the On/Off button). Depth & Time/ Sweep Knobs The function and label of the second and third controls in the Mod Delay module are determined by the Mode pull-down menu.
Time Knob Sets the modulation delay time. The available range is from 0 to 125 milliseconds. In Vibrato mode, this setting will appear to have no effect if the Recirculation value is zero because the signal is “100% wet” in Vibrato mode. Note: Recirculation (Recir) Knob The Time knob is only visible in Chorus and Vibrato modes. Sets the amount of processed signal fed back into its input. Higher values increase the intensity of the processed signal. Recirculation allows both positive and negative values.
Mod Delay LFO Type Table Mode Menu Table 32. Mod Delay LFO Types and Descriptions Sin 0 In-phase sine wave Sin 90 Sine wave 90 degrees out of phase Sin 180 Sine wave 180 degrees out of phase Tri 0 In-phase triangle wave Tri 90 Sine wave 90 degrees out of phase Tri 180 Sine wave 180 degrees out of phase Trem Up The Trem/Fade module is used as the LFO source. On a stereo signal, both channels ascend in pitch in synchronization with the Trem/Fade amplitude ramp.
Echo Module Figure 138. The Echo module The Echo module is a delay line used primarily for longer echo effects. When very short delay times or modulation are desired, use the Mod Delay instead. When VERY long delay times are desired, use the UAD DM-L plug-in which has up to 2400 milliseconds available delay per stereo channel. Sync Button This button puts the plug-in into Tempo Sync mode. See “Tempo Sync” on page 96 for more information.
Damping Knob This low pass filter reduces the amount of high frequencies in the processed signal. Higher values yield a brighter signal. Turn down this control for a darker sound. Damping also mimics air absorption, or high frequency rolloff inherent in tape-based delay systems. Mix Knob This control determines the balance between the delayed and original signal. Values greater than 50% emphasize the wet signal, and values less than 50% emphasize the dry signal.
CHAPTER 39 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 139. 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. See “Text Entry” on page 92 for more information. 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 40 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. See “Text Entry” on page 92 for more information. 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 41 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. See “Text Entry” on page 92 for more information. 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 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 42 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 142. 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. See “Text Entry” on page 92 for more information. The Effect Knob controls the amount of processing that occurs in the plug-in. The available range is from 0.00 to 100.0%.
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 43 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” on page 74 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 “Compensating Upsampled Plug-Ins” on page 108 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. See “Host PDC Implementation” on page 101.
CHAPTER 44 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 144. 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. See “Text Entry” on page 92 for more information. 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 451). PK-RMS K-20 K-14 K-12 Figure 145. 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 146. 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 45 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 147. 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. See “Text Entry” on page 92 for more information. 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” on page 74 is enabled). Note: The Limit function provides a second stage of soft-saturation just before the output control for the plug-in.
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.
Compensating for Precision Maximizer 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. See “Host PDC Implementation” on page 101. Note: WebZine Article An interesting article about sonic enhancers can be found in the “Ask The Doctors” article of the Universal Audio May 2007 Webzine (Volume 5, Number 4), published on the internet at: • http://www.uaudio.com/webzine/2007/may/index2.
CHAPTER 46 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 470). 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 463). 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 149. 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 468). 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 (see “Text Entry” on page 92).
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” on page 74 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 47 Pultec and Pultec-Pro Overview The Pultec EQP-1A Program Equalizer and Pultec MEQ-5 plug-ins are faithful electronic reproductions of the classic hardware equalizers. Our DSP wizards have ensured that every revered sonic nuance of these vintage processors are faithfully maintained. UAD Pultec and UAD Pultec-Pro The UAD Pultec plug-in is the EQP-1A Program Equalizer that was introduced in version 2.2 to much acclaim. UAD Pultec-Pro was introduced in version 3.
Pultec Latency The Pultec and Pultec-Pro plug-ins introduce an additional 13 samples of delay due to upsampling when the session sample rate is below 100 kHz. This additional latency does not occur at sample rates above 100 kHz. You may enter a value of 13 in the “Samples” parameter in DelayComp or TrackAdv to compensate. See “Compensating Upsampled Plug-Ins” on page 108 for more information.
Figure 151. Control grouping within the Pultec EQP-1A In/Out Toggle Switch This is a signal bypass control. It allows you to compare the processed and unprocessed signal. It does NOT reduce UAD DSP load. In the hardware EQP-1A, the audio is still slightly colored even when the switch is in the Out position. This is due to the fact that the signal is still passing through its circuitry.
Note: 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. In actual use however, the Boost control has slightly higher gain than the Attenuation has cut, and the frequencies they affect are slightly different too. The EQ curve that results when boost and attenuation are simultaneously applied to the low shelf is an additional feature.
Pultec MEQ-5 Screenshot Figure 152. The Pultec-Pro MEQ-5 Midrange Equalizer plug-in window Pultec MEQ-5 Controls The MEQ-5 can control three frequency ranges simultaneously, using three groups of interacting parameters. The first group controls the low–mid frequencies and has two controls: frequency select and boost. The second group controls the mid frequencies and has two controls: frequency select and attenuation. The third group controls high-mids and has two controls: frequency select and boost.
In the hardware MEQ-5, the audio is still slightly colored even when the switch is in the Out position and the peak/dip controls are at zero. This is due to the fact that the signal is still passing through its circuitry. Because the plugin emulates the hardware in every regard, the signal will be slightly processed when this switch is in the In position and the peak/dip controls are at zero. If a true bypass is desired, use the host disable switch.
Low Peak Response Figure 154.
Dip Response Figure 155.
High Peak Response Figure 156.
CHAPTER 48 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 157. RealVerb Pro signal flow Figure 157 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 158). 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 160. 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 162 on page 493). 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 161.
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 162. 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.
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. For a full stereo signal, drag the left handle all the way to left, and right handle all the way to the right. 2. Drag the blue center of the slider left or right to set the positioning of the sig- nal. If you drag all the way to the left or right, the stereo width is adjusted.
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. This approach is in contrast to the traditional method of morphing by crossfading between the output of two static reverberators. The method employed by RealVerb Pro produces more faithful, physically meaningful intermediate states. Figure 165.
Figure 166.
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 49 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 Stereo Mode switch determines the operating mode of CE-1 when the plug-in is used in a configuration with stereo input, such as a stereo audio track insert or stereo effects bus. Stereo Mode Switch 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.
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 50 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 51 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 169. 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 41 on page 508. Table 41. 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 507.
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 52 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 519). 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 172 on page 520 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 172.
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 176 on page 522 displays the processed waveforms with maximum and minimal sustain to compare against the original waveform in diagram 4. Figure 176. 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 53 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 177. The UAD SSL E and 4K Channel Strip plug-in windows SSL E Channel Strip Controls The SSL E Channel Strip controls are divided into four main sections: filters, dynamics, EQ, and global. Note: Knob settings, when compared to the graphical user interface 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 530 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 529). 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.ATT” LED 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 red 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 54 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 55 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 179. 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 180) 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 546 and “Secondary Controls” on page 549. Figure 180.
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 551 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 550) 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 181. 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 550). 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.
The Studer A800 Professional Multichannel Magnetic Tape Recorder All visual and aural references to Studer products and all use of Studer trademarks are being made with written permission from Harman International Industries, Inc. Any references to third party tape formulations are used solely for identification and do not imply any endorsement by, or affiliation with, any tape manufacturer.
CHAPTER 56 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 183). 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 57 History Bill Putnam Sr. The name M.T. “Bill” Putnam retains a unique status in the audio industry hall of fame- it's legendary even among those who are considered to be legends themselves. Called the “father of modern recording” by no less a luminary than Bruce Swedien, and a “visionary, responsible for motivating new thinking,” by respected studio engineer Tom Hidley, Putnam was a true renaissance man in the world of sound and music.
Universal Audio. It wasn't long before the company relocated to Chicago, and it was there, in 1947, that Putnam recorded what is generally accepted to be the first “pop” record to use artificial reverberation. The founder of the group The Harmonicats, Jerry Murad, wanted to record using an echo chamber like he'd heard on effects in spooky radio mysteries.
With all this success, Universal Recording went through several incarnations, with the dream version completed in 1955. At that time it was the most advanced and largest independent recording facility in the country attracting top producers like Nelson Riddle, Mitch Miller and Quincy Jones. It was also at that time that Bruce Swedien went to work for the studio. “It's absolutely true,” he states. “Bill Putnam was the father of recording as we know it today.
well as Sinatra's “It Was A Very Good Year,” and The Mamas and The Papas' “California Dreamin'”. The United Western studios, still in existence today as both Cello Studios and Allen Sides' Ocean Way Recording, are still considered to be some of the best sounding rooms ever built. Universal Audio and UREI Meanwhile, upstairs in the 6050 Sunset building Universal Audio was thriving, and changing names.
he started recording everything with feeds to two control rooms, one for a stereo mix, one for a mono mix. In late '58, '59 and '60 everything Bill did was recorded in both stereo and mono. “When stereo hit big around '61, none of the record companies had any catalog. But Bill did — he had two and one half years worth. It was a lot of material — understand at that period of time he was doing about $200,000 a month in the United Western Complex — which is like a million dollars a month now.
The basic concept of a compressor/limiter, is of course, relatively simple. It's a device in which the gain of a circuit is automatically adjusted using a predetermined ratio that acts in response to the input signal level. A compressor/limiter “rides gain” like a recording engineer does by hand with the fader of a console: it keeps the volume up during softer sections and brings it down when the signal gets louder.
After several unsuccessful attempts at using F.E.T.s in gain reduction circuits, Putnam settled upon the straightforward approach of using the F.E.T. as the bottom leg in a voltage divider circuit, which is placed ahead of a preamp stage. The output stage of the 1176 is a carefully crafted class A line level amplifier, designed to work with the (then) standard load of 600 ohms. The heart of this stage is the output transformer, whose design and performance is critical.
Pros Talk UA Both the 1176 and the LA-2A remain in daily use. Busy engineers and producers’ comments about both the 1176 and the LA-2A demonstrate their impact on the industry: Mike Shipley: Mike Shipley (Def Leppard, Shania Twain, Blondie): “I grew up using 1176s – in England they were the compressor of choice. They're especially good for vocals, which is also what I primarily use the LA-2 for. Most anything else I can do without, but I can't be without at least a pair of 1176s and an LA-2A.
and string dates. Among his recent credits are work with the Goo Goo Dolls, Alanis Morissette and Green Day. Sides brings his different perspectives into play when he talks about using the 1176. “The 1176 is standard equipment for my sessions. I just used them last night, as a matter of fact, on a project for singer Lisa Bonet that Rob Cavallo was producing at Ocean Way. We were recording drums and I used them on the left/right overheads as effects limiters.
Jim Scott Jim Scott shared a Grammy for Best Engineered Album for Tom Petty's Wildflowers. He's also known for his work with Red Hot Chili Peppers, Natalie Merchant and Wilco. “I use 1176s real conservatively and they still do amazing things,” he comments. “I'm always on the four to one button, and the Dr. Pepper – you know, 10 o'clock, 2 o'clock, and it does everything I need. “I always use them on vocals.
“My big mentors were Andy Johns and Lee DeCarlo and Ron Nevision because they were all Record Plant guys. I learned how to make a rock and roll record from them. Although over the years it's become my own thing, my style still tends to be that Record Plant style, U87s, 1176s, LA-2As, 47 F.E.T.s...it's what I like.” Mike Clink Producer/Engineer Mike Clink (Guns N' Roses, Sammy Hagar, Pushmonkey) also comes from the Record Plant school of recording.
Thank You We would like to thank you again for becoming a Universal Audio customer. We urge you subscribe to our email lists so we can keep you informed about UA product developments and promotions. Email preferences are set in your registration profile (see “My Profile” on page 53). We always like to hear from our customers and welcome your comments and suggestions. If you have any questions you can email us at: • info@uaudio.
INDEX Numerics 1176LN 281, 568 1176LN Controls 287 1176SE “Special Edition” 289 1176SE Controls 289 4K Buss Compressor 537 A A/B Selector 176, 375, 443, Accessing Meter Functions 63 Account 53 acoustical space 209 Adjusting Parameters 87, 91 AGC Mode 267 Air Blending 211 Air Density Menu 210 algorithm 208 All Button 464 All Buttons mode 288 Always On Top 62 AMD-8131 Mode 75 Amp Bent Knob 400 Amp Color Knob 400 Amp EQ Groups 399 Amp On/Off Button 401 Amp Output Knob 400 Amp Overview 398 Amp Post-EQ Knobs 40
INDEX Boss CE-1 Controls 500 Boss CE-1 Screenshot 499 Bright Button 401 Buy Button 71, 91 Buying Plug-Ins 53 Bypass 212, 213 C Cabinet 402 Cabinet On/Off Button 404 Cabinet Type Menu 402 Cambridge EQ Controls 174 Cambridge EQ Screenshot 173 Cambridge Equaliser 173 Card Info Display 59 Card Status 67 CE-1 Chorus Intensity Knob 501 CE-1 Clip LED 500 CE-1 Depth Knob 502 CE-1 Normal/Effect Switch 500 CE-1 Output Level Knob 501 CE-1 Rate Knob 502 CE-1 Rate LED 500 CE-1 Stereo Mode Switch 501 CE-1 Vibrato Contro
INDEX Depth Knob 195, 413 Desktop 62 Detailed System Profile 68 Diffusion 214 Dimension D 503 Dimension D Controls 504 Dimension D Screenshot 503 Dimension Mode 504 Disable Current 63 Disabling Cards 59 Disconnect 125, 134 Distance 216 DM-1 Controls 194 DM-1 Delay Modulator 194 DM-1L 194 DM-1L Overview 196 Documentation Overview 21 DreamVerb 203 Dry 217 DSP 64 DSP Load 68 DSP Load Limiting Overview 72 DSP Loading Information 93 DSP LoadLock 74 DSP Settings 84 Dual Mode 442, 501 Dynamics 379 Dynamics Meters
INDEX Features 20 Helios Type 69 Equalizer 275 Feedback 24 Helios Type 69 Screenshot 275 Filter Type Menu 410 Help & Support Panel 84 Filtering 212 Help Menu 71, Firewire Bandwidth vs.
INDEX L LA-2A 281, 567 LA-3A Compressor 290 LA-3A Controls 291 LA-3A Screenshot 290 Late 215 Late-Field Relative Timing 213 Late-Field Start 214 Latency 66 latency 101 Latency & Delay Compensation 101 Latency Calculator 84 Latest Information 27 Launching a Powered Plug-In 85 Launching a UAD Powered Plug-In 85 Launching the Meter Windows 62 Launching the UAD Meter & Control Panel 62 L-Delay Knob 194 Levels 216 Lexicon 224 292 Lexicon 224 Controls 299 Lexicon 224 Screenshot 294 LFO Mode 409 LFO Type Menu 4
INDEX Mix 217, 457 Mix Knob 239, 405, 420 Mixed UAD System 55 Mod Delay On/Off Button 418 Mod Delay Overview 415 Mod Depth 238 Mod Filter Module 407 Mod Filter On/Off Button 410 Mod Rate 238 Mod Type Menu 409 Mode Menu 413 Mode Pop-up Menu 195 Mode Switch 448 Modes 441 Modulation 238 Moog Filter Controls 333 Moog Filter Latency 340 Moog Filter Screenshot 333 Moog Filter SE 339 Moog Filter SE Controls 339 Moog Multimode Filter 332 Morph Slider 402 Multicard DSP Loading 58 Multicard Setup 120 Multiple Buttons
INDEX Overview 190, 200, 275, 290, 349, 354, 361, 367, Precision Enhancer Hz Usage Notes 435 377, 421, 427, 431, 436, 454, 461, 505, 512 Precision Enhancer kHz 436 Precision Enhancer kHz Controls 437 P Package Contents 21, 116, 129 parameter 91 Parameter Copy Buttons 442 Parametric EQ 179 Parametric Type Selector 179 Peak 291 Peak Level 350, 355, 362, 506 Peak Reduction 285 Phasor Module 404 Phasor On/Off Button 406 Plate 140 Overview 234 Plate Select Switch 236 Platforms 22 Plug-In Formats 45 Plug-I
INDEX Purchasing Plug-Ins 53 Q Q (Bandwidth) Knob 179 R Rate 408 Rate Knob 195, 404, 408, 415 Rate knob 413 Ratio 287 Ratio Knob 193 R-Delay Knob 194 ReadMe 21 RealVerb Pro 484 Roland CE-1 Overview 499 Roland Dimension D 503 Roland Dimension D Controls 504 Roland Dimension D Overview 503 Roland Dimension D Screenshot 503 Roland RE-201 505 Roland RE-201 Controls 506 Roland RE-201 Interface 506 Roland RE-201 Screenshot 506 Room Shape and Material 484 R-Pan Knob 196, 199 RS-1 Controls 198 RS-1 Reflection
INDEX Size Knob 198 System Information Panel 65 Slope 214 System Overview 38 Software Installation 28 System Requirements 26, Software Instructions 124 Software Removal 37 T Software Updates 27 Target Link Speed 80 Solo 428 Tempo Sync 96 Solo Display 472 Tempo Sync Modes 99 Space 219 Tempo Sync Plugins 96 Spatial Characteristics 218 Spectral Characteristics 486 spectral characteristics 213 Speed 428 SPL Transient Designer 512 SPL Transient Designer Applications 515 SPL Transient Designer Con
INDEX UAD 116, 129 UAD Bandwidth Allocation 80 UAD Control Panel 64 UAD Drivers 43 UAD Environment 38 UAD ExpressCard Products 114 UAD Hardware 31, 38, 120 UAD Meter & Control Panel 43, 61 UAD Nomenclature 22 UAD Plug-In Window 86 UAD Software 42 UAD Software Installation 28 UAD Toolbar 87 UAD-1 Family 40 UAD-1 Hardware 31, 120 UAD-1 Host Compatibility 75 UAD-1 PCI 74 UAD-2 DSP 73 UAD-2 Family 39 UAD-2 LED 36 UAD-2 Satellite 128 UAD-2 Satellite Notes 135 UAD-2 Satellite Operation 133 UAD-2 SOLO/Laptop 116,
INDEX X Xpander Connections 123 Xpander Notes 127 Xpander Operation 123 Xpander System Sleep 126, 135 Z Zoom Buttons 175 UAD Powered Plug-Ins Manual - 585 - Index
