E-MU 1616 v2 PCI, 1212M v2 PCI Owner`s manual

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Owner’s ManualOwner’s Manual
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E-MU 1616/1616m/1212m PCI Digital Audio System
Owner’s Manual
All Rights Reserved
Software Version: 1.82
E-MU World Headquarters
E-MU Systems
1500 Green Hills Road
Scotts Valley, CA 95066
USA
Asia Pacific, Africa, Middle East
Creative Technology Ltd
31 International Business Park
Creative Resource. Singapore 609921
SINGAPORE
2 Creative Professional
Europe
Creative Labs
Ballycoolin Business Park
Blanchardstown
Dublin 15
IRELAND
E-MU Japan
Creative Media K K
Kanda Eight Bldg., 3F
4-6-7 Soto-Kanda
Chiyoda-ku, Tokyo 101-0021
JAPAN
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Table of Contents
1- Introduction ................................................................. 7
Welcome!.............................................................................................................................. 7
All Systems Include: ......................................................................................................... 9
E-MU 1212M System ....................................................................................................... 9
E-MU 1616 System .......................................................................................................... 9
E-MU 1616M System ....................................................................................................... 9
PatchMIx DSP ................................................................................................................. 10
Notes, Tips and Warnings ..........................................................................................10
2 - Installation ................................................................ 11
Setting Up the Digital Audio System ................................................................................ 11
Notes for Installation ................................................................................................. 11
Safety First! ................................................................................................................. 12
Connector Types ............................................................................................................ 12
Installing the E-MU 1010 PCI Card .................................................................................. 13
Install the Sync Daughter Card or 0202 Daughter Card.................................................. 14
WARNING : E-MU 0202 & MicroDock .................................................................... 14
Connecting the MicroDock ...............................................................................................15
Software Installation .......................................................................................................... 16
Installing the E-MU 1010 Drivers .............................................................................16
Windows XP ...............................................................................................................16
Uninstalling all Audio Drivers and Applications ..................................................... 16
Note About Windows Logo Testing .......................................................................... 16
3 - PCI Card & Interfaces ................................................. 17
The E-MU 1010 PCI Card.................................................................................................. 17
Important .................................................................................................................... 17
Connections ................................................................................................................... 17
EDI Connector ............................................................................................................ 17
S/PDIF Digital Audio Input & Output ...................................................................... 17
ADAT Optical Digital Input & Output ......................................................................17
The 0202 Daughter Card ................................................................................................... 18
Connections ................................................................................................................... 18
Analog Inputs and Outputs ....................................................................................... 18
MIDI In/Out ............................................................................................................... 18
E-MU PCI Digital Audio Systems 3
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The MicroDock...................................................................................................................19
Front Panel Connections ............................................................................................... 20
Preamp Section ........................................................................................................... 20
S/PDIF Digital Audio Input & Output ...................................................................... 20
ADAT Optical Digital Input & Output ......................................................................21
Headphone Output & Volume Control .................................................................... 21
Rear Panel Connections ................................................................................................. 23
Line Level Analog Inputs ...........................................................................................23
Turntable Inputs & Ground Lug ................................................................................23
Line Level Analog Outputs ........................................................................................23
Computer Speaker Analog Outputs .......................................................................... 24
MIDI 1 & 2 In/Outs .................................................................................................... 24
EDI Connector (Card) ...............................................................................................24
4 - The PatchMix DSP Mixer ............................................. 27
PatchMix DSP..................................................................................................................... 27
Overview of the Mixer........................................................................................................ 27
Mixer Window ................................................................................................................ 28
Mixer Block Diagram ..................................................................................................... 28
Pre Fader or Post Fader ..............................................................................................28
E-MU Icon in the Windows Taskbar ................................................................................. 29
The Toolbar ........................................................................................................................ 29
The Session ......................................................................................................................... 30
New Session ....................................................................................................................30
Open Session .................................................................................................................. 31
Save Session .................................................................................................................... 31
Session Settings .............................................................................................................. 31
System Settings ........................................................................................................... 31
Using External Clock .................................................................................................. 32
I/O Settings ................................................................................................................. 32
Input Mixer Strips............................................................................................................... 34
Input Type ................................................................................................................... 34
Mixer Strip Creation........................................................................................................... 35
Multichannel WAVE Files .............................................................................................. 36
Windows Media Player/DVD/Surround Sound Playback ....................................... 36
Insert Section .................................................................................................................. 37
Working with Inserts ..................................................................................................37
The Insert Menu ......................................................................................................... 38
ASIO Direct Monitor Send/Return ............................................................................ 39
Meter Inserts ............................................................................................................... 40
To Set the Input Levels of a Strip ................................................................................... 41
Making the Best Possible Recording ......................................................................... 41
Trim Pot Insert ............................................................................................................ 42
Test Tone/Signal Generator Insert ............................................................................. 43
Managing Your Inserts ................................................................................................... 44
Aux Section ..................................................................................................................... 45
Sidechain Diagram ..................................................................................................... 45
Pre or Post Fader Aux Sends ......................................................................................46
Level, Pan, Solo & Mute Controls ................................................................................. 47
Main Section....................................................................................................................... 48
TV Screen & Selectors ..................................................................................................... 49
Effect ........................................................................................................................... 49
Input ........................................................................................................................... 50
Output ........................................................................................................................ 50
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Auxiliary Effects & Returns ............................................................................................ 51
Sidechain Diagram ..................................................................................................... 51
Sync/Sample Rate Indicators ......................................................................................... 51
Output Section ............................................................................................................... 52
Main Inserts ................................................................................................................ 52
Main Output Fader ..................................................................................................... 52
Output Level Meters ...................................................................................................52
Monitor Output Level ................................................................................................ 52
Monitor Balance Control ...........................................................................................52
Monitor Output Mute ................................................................................................ 52
5 - Effects ....................................................................... 53
Overview.............................................................................................................................53
The Effects Palette............................................................................................................... 53
FX Insert Chains ............................................................................................................. 54
Creating, Renaming & Deleting Categories or Presets ............................................. 55
Importing and Exporting Core FX Presets and FX Insert Chains ............................. 56
FX Edit Screen.....................................................................................................................57
User Preset Section ......................................................................................................... 58
Core Effects and Effects Presets .....................................................................................59
List of Core Effects.............................................................................................................. 60
DSP Resource Usage ....................................................................................................... 60
Core Effects Descriptions................................................................................................... 61
1-Band Para EQ .............................................................................................................. 61
1-Band Shelf EQ ............................................................................................................. 61
3-Band EQ ......................................................................................................................62
4-Band EQ ......................................................................................................................63
Auto-Wah ........................................................................................................................ 64
Chorus ............................................................................................................................ 65
Compressor .................................................................................................................... 65
Basic Controls ............................................................................................................. 66
Distortion ....................................................................................................................... 67
Flanger ............................................................................................................................ 68
Freq Shifter .....................................................................................................................69
Leveling Amp .................................................................................................................. 70
Lite Reverb ...................................................................................................................... 71
Mono Delays - 100, 250, 500, 750, 1500, 3000 .......................................................... 72
Phase Shifter ................................................................................................................... 73
Rotary .............................................................................................................................. 73
Speaker Simulator .......................................................................................................... 74
Stereo Delays - 100, 250, 500, 750, 1500 .................................................................... 75
Vocal Morpher ................................................................................................................ 77
E-MU PowerFX ................................................................................................................... 78
Automating E-MU PowerFX .......................................................................................... 80
E-MU PowerFX Resource Availability ........................................................................... 80
Rendering Audio with E-MU PowerFX .............................................................................82
General Tips for Rendering using PowerFX .............................................................. 82
Tips for using Freeze Mode on Cubase LE ................................................................ 82
Using E-MU PowerFX with WaveLab and SoundForge ...............................................82
E-MU VST E-Wire ...............................................................................................................83
E-Delay Compensator ....................................................................................................84
E-Delay Compensator Use ......................................................................................... 85
E-Delay Units Parameter ............................................................................................ 85
Grouping Tracks ......................................................................................................... 86
E-MU PCI Digital Audio Systems 5
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6 - Appendix ................................................................... 87
Using High Sample Rates .................................................................................................. 87
Overview ......................................................................................................................... 87
WDM Recording and Playback Behavior ...................................................................... 89
Sync Daughter Card Supplement...................................................................................... 90
SMPTE Conversion ........................................................................................................ 90
SMPTE Features .......................................................................................................... 90
SMPTE Options .............................................................................................................. 90
SMPTE Modes of Operation .......................................................................................... 91
Host Mode .................................................................................................................. 91
External Mode ............................................................................................................ 91
Flywheel Mode ........................................................................................................... 91
Flywheel Modes .......................................................................................................... 91
Stripe Mode ................................................................................................................ 92
Example SMPTE Connection ......................................................................................... 92
MIDI Time Code (MTC) .................................................................................................... 93
Word Clock In/Out ............................................................................................................ 93
Getting in Sync ................................................................................................................... 95
Useful Information ............................................................................................................96
Cables - balanced or unbalanced? ................................................................................ 96
Balanced Cables ......................................................................................................... 96
Unbalanced Cables ....................................................................................................96
Adapter Cables ...............................................................................................................97
1/8ā€ Mini-phone to 1/4ā€ Adapters ............................................................................ 97
Cinch (RCA) to 1/4ā€ Adapters ................................................................................... 97
Digital Cables .................................................................................................................97
AES/EBU to S/PDIF Cable Adapter ...............................................................................97
Grounding ......................................................................................................................98
Phantom Power .............................................................................................................. 98
Appearance Settings in Windows .................................................................................. 98
Technical Specifications.....................................................................................................99
Internet References...........................................................................................................107
Forums ...................................................................................................................... 107
Index ............................................................................ 111
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•
•
•

1- Introduction

Welcome!

Thank you for purchasing the E-MU 1616 PCI, E-MU 1616M PCI, or 1212M PCI Digital Audio System. Your computer is about to be transformed into a powerful audio processing workstation. We’ve designed this E-MU digital audio system to be logical, intuitive and above all, to provide you with pristine sound quality. These systems offer unprecedented quality and value by providing studio-class, 24-bit/192kHz multiĀ­channel recording and playback to any PCI card bus equipped PC.
1- Introduction
Welcome!
1616 PCI System Components
E-MU 1616 PCI
• E-MU 1010 PCI Card
• MicroDock
• EDI (E-MU Digital Interface Cable)
• +48VDC AC Adapter
MIDI Breakout Cable Digital Audio System Software/Driver Installation CD-ROM Production Tools Software Bundle CD-ROM
• Quick Start Guide
Inputs & Outputs
(8) Channel ADAT Digital Optical Input (8) Channel ADAT Digital Optical Output (2) Channel S/PDIF Digital Input (2) Channel S/PDIF Digital Output (2) MIDI Inputs & Outputs (4) 24-bit Balanced Line Inputs (6) 24-bit Balanced Line Outputs (2) Microphone/Line Preamp Inputs (2) Turntable Preamp Inputs (1) Stereo Headphone Output (3) Stereo Computer Speaker Outputs
(allows 32 MIDI channels)
(with +48V phantom power)
(with RIAA equalized preamplifier)
(with volume control)
(with 1/8ā€ jacks to connect powered speakers)
E-MU PCI Digital Audio Systems 7
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•
•
•
•
1- Introduction Welcome!
1616M PCI System Components
E-MU 1616M PCI
• E-MU 1010 PCI Card
• MicroDockM
• Sync Card
• EDI (E-MU Digital Interface Cable)
• +48VDC AC Adapter
MIDI Breakout Cable Digital Audio System Software/Driver Installation CD-ROM Production Tools Software Bundle CD-ROM
• Quick Start Guide
Inputs & Outputs
(8) Channel ADAT Digital Optical Input (8) Channel ADAT Digital Optical Output (2) Channel S/PDIF Digital Input (2) Channel S/PDIF Digital Output (2) MIDI Inputs & Outputs
(allows 32 MIDI channels)
(4) 24-bit Balanced Line Inputs (6) 24-bit Balanced Line Outputs (2) Microphone/Line Preamp Inputs (2) Turntable Preamp Inputs
(with RIAA equalized preamplifier)
(1) Stereo Headphone Output (3) Stereo Computer Speaker Outputs
(with +48V phantom power)
(with volume control)
(with 1/8ā€ jacks to connect powered speakers)
(1) Word Clock In & Out (1) SMPTE (LTC) In & Out
1212M PCI System Components
E-MU 1212M
• E-MU 1010 PCI Card
• E-MU 0202 I/O Daughter Card
• 0202 I/O Card Cable
• (2) MIDI Adapter Cables
• Digital Audio System Software/Driver Installation CD-ROM
Production Tools Software Bundle CD-ROM
• Quick Start Guide
Inputs & Outputs
(8) Channel ADAT Digital Optical Input (8) Channel ADAT Digital Optical Output (2) Channel S/PDIF Digital Input (2) Channel S/PDIF Digital Output (1) MIDI Input & Output
(allows 16 MIDI channels)
(2) 24-bit Balanced Line Inputs (2) 24-bit Balanced Line Outputs
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All Systems Include:

The E-MU 1010 PCI Card is the heart of all three systems. Its powerful hardware DSP processor allows you to use over 16 simultaneous hardware-based effects, which place minimal load on your computer’s CPU. The E-MU 1010 PCI Card also provides eightĀ­channels of ADATĀ® optical digital input and output, as well as a S/PDIF stereo digital input and output.
The PatchMix DSP mixer application is included in all the systems. PatchMix DSP delivers unmatched flexibility in routing your audio between physical inputs and outputs, virtual (ASIO/WAVE) inputs and outputs and internal hardware effects and buses—no external mixer needed. You can add digital effects, EQs, meters, level controls and ASIO/WAVE sends anywhere you like in the signal chain.
Because the effects and mixing are hardware-based, there is no latency when you record. You can even record a dry signal while monitoring yourself with effects! Mixer setups can be saved and instantly recalled for specific purposes such as recording, mixdown, jamming, special effect setups, playing games, watching DVDs, or general computer use.
1- Introduction
Welcome!

E-MU 1212M System

The E-MU 1212M includes the 0202 Daughter Card, which provides 2 line level, balanced analog inputs, 2 line level, balanced analog outputs, plus MIDI input and output. This is no-compromise audio interface, using ultra-high performance 24-bit/192kHz A/D - D/A converters to deliver an unbelievable 120dB dynamic range.

E-MU 1616 System

The E-MU 1616 includes the MicroDock, which is a half rack-space, audio interface. The MicroDock adds the following input and output capabilities to the system: two mic/line inputs with custom low-noise preamps, 4 balanced line level analog inputs, an RIAA stereo turntable preamp, 6 balanced line level outputs, an assignable headphone output, two sets of MIDI I/O ports, an additional S/PDIF optical output, and four stereo mini phone jacks for easy connection to powered speaker systems. Of course, profesĀ­sional-quality, 24-bit A/D and D/A converters with automatic DC blocking are used throughout.

E-MU 1616M System

The E-MU 1616M system includes the MicroDockM, a no compromise, mastering-grade system, which includes all the features of the 1616 system. The 1616M system is distinĀ­guished by the addition of ultra-high performance 24-bit/192kHz A/D - D/A converters which deliver an incredible 120dB of dynamic range.
The Sync Daughter Card comes standard with the 1616M system and can be purchased as an optional upgrade for the 1616 MicroDock and 1212M systems. The Sync Card adds Word Clock in and out for sample-synchronizing outboard digital equipment and SMPTE longitudinal time code in/out for syncing other recording equipment. A separate MIDI Time Code output port on the Sync Card eliminates timing problems caused by combining MTC with MIDI performance data.
E
Note: The digital I/O on the 1010 PCI card are not used when you have the MicroDock.
You’ll want to keep up with the latest software and options for your E-MU digital audio system. You can find all of this, plus other helpful information, at the E-MU Website: http://www.emu.com.
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1- Introduction Welcome!

PatchMIx DSP

PatchMix DSP offers unmatched flexibility in routing your audio between physical inputs/outputs, virtual (ASIO/WAVE) inputs/outputs, internal hardware effects and buses. No external mixer is needed. You can add digital effects, EQs, meters, level controls and ASIO/WAVE sends anywhere you like in the signal chain.
Because the effects and mixing are hardware-based, you can record using effects with near zero-latency. You can even record a dry signal while monitoring yourself with effects! Mixer setups can be saved and instantly recalled for specific purposes such as recording, mixdown, jamming, special effect setups, playing games, watching DVDs, or general computer use.
You’ll want to keep up with the latest software and options for your E-MU digital audio system. You can find all of this, plus other helpful information, at the E-MU Website: http://www.emu.com.
Notes, Tips and Warnings
Items of special interest are presented in this document as notes, tips and warnings.
f Notes provide additional information related to the topic being discussed. Often,
notes describe the interaction between the topic and some other aspect of the system.
E Tips describe applications for the topic under discussion.

Warnings are especially important, since they help you avoid activities that can
cause damage to your files, your computer or yourself.
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2 - Installation

Setting Up the Digital Audio System

There are six basic steps to installing your E-MU system:
Remove any other sound cards you have in your computer. (Once you are sure that
1.
the E-MU card works properly, your old sound card can be reinstalled if desired.)
Install the E-MU 1010 PCI card in your computer. Go there.
2.
Install the 0202 Daughter Card or Sync Daughter Card (if applicable). Go there.
3.
4.
Connect the MicroDock (if applicable).
5.
Install the PatchMix DSP software onto your computer.
Connect audio, MIDI and synchronization cables between the E-MU system and
6.
your other gear.

Notes for Installation

2 - Installation
Setting Up the Digital Audio System
• IF AT ANY TIME DURING THIS INSTALLATION YOU SEE NO RESPONSE: Use the Alt-Tab feature to select other applications. One of them may be the Microsoft Digital Signature warning. It is possible for this warning to appear behind the installation screen.
•Make sure you have the latest Windows Service Packs from Microsoft (Windows XP - SP 1 or higher).
• Disable onboard sound and uninstall all other sound cards. (If you wish to try using multiple sound cards in your system, do so after you have confirmed that your E-MU Digital Audio System is operating normally.)
• InstallShield ā€œIKernel Application Errorā€ on Windows XP: When installing this software on Windows XP, you may be confronted with a ā€œkernel errorā€ at the very end of installation. This is an issue with the InstallShield program, which is what we use to install software on your computer. Please do not be alarmed by this, as the error is innocuous.
To read more about this error, and obtain instructions on how to avoid getting the message, please visit this website: http://support.installshield.com/kb/view.asp?articleid=q108020
•Multiple Digital Audio System sound cards are not supported.
E-MU PCI Digital Audio Systems 11
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2 - Installation Setting Up the Digital Audio System
Please read the following sections as they apply to your system as you install the E-MU 1010, paying special attention to the various warnings they include.
Prior to installing the hardware, take a few moments to write down the 18-digit serial number, which is located on the back of the box and on the 1010 PCI Card. This number can help EMU Customer Service troubleshoot any problems you may encounter—by writing the number down now, you’ll avoid having to open your computer to find it later on.

Safety First!

• To avoid possible permanent damage to your hardware, make sure that all connecĀ­tions are made with the host computer’s power off. Unplug the computer’s
power cable to make sure that the computer is not in sleep mode .
• Take care to avoid static damage to any components of your system. Internal computer surfaces, the E-MU 1010 PCI board and the interfaces are susceptible to electrostatic discharge, commonly known as ā€œstatic.ā€ Electrostatic discharge can damage or destroy electronic devices. Here are some procedures you can follow when handling electronic devices in order to minimize the possibility of causing electrostatic damage:
As you install

hardware components, observe the following general precautions to avoid damage to your equipment and yourself.
• Avoid any unnecessary movement, such as scuffing your feet when handling electronic devices, since most movement can generate additional charges of static electricity.
• Minimize the handling of the PCI card. Keep it in its static-free package until needed. Transport or store the board only in its protective package.
• When handling a PCI card, avoid touching its connector pins. Try to handle the board by its edges only.
• Before installing a PCI card into your computer, you should be grounded. Use a ground strap to discharge any static electric charge built up on your body. The ground strap attaches to your wrist and any unpainted metal surface within your computer. If you don’t have a ground strap, you can ground yourself by touching the metal case of another piece of grounded equipment.
• Before connecting a cable to your interface or between PCI cards, touch the connector sleeve of the cable to the sleeve of the jack to which you’ll be connecting the cable in order to discharge any static build-up.

Connector Types

These connector types are used to connect the E-MU 1010 hardware components. They will be referred to by the name shown in the first column of the following chart:
Name Description Connects
Card/External CAT5 Connector 1010 PCI card and MicroDock
S/PDIF In RCA Connector S/PDIF digital audio devices
S/PDIF Out RCA Connector S/PDIF digital audio devices
ADAT Optical In TOSLINK Optical Connector ADAT digital audio devices (or S/PDIF)
ADAT Optical Out TOSLINK Optical Connector ADAT digital audio devices (or S/PDIF)

Warning: Please verify that all cables are connected only to the proper components
before powering up your system.
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Installing the E-MU 1010 PCI Card

Note: This installation is very simple but if you are not familiar with the installation
of computer peripherals and add-in boards, please contact your authorized E-MU Systems dealer or an approved computer service center to arrange for the installation.
To install the 1010 PCI card into your computer
ī˜‚
1.
Make sure that the power switch on your computer is off.
IMPORTANT: Unplug the power cord from the wall outlet!
2.
Touch a metal plate on your computer to ground yourself and to discharge any static electricity.
Follow the computer manufacturer’s recommended procedure for opening the case.
3.
4.
Remove the metal brackets from two adjacent PCI slots. If you have the E-MU 1820 system (non-M) you only need to remove the bracket from a single PCI slot. Put the screw(s) aside for use later. See figure 1 below.
Figure 1 Figure 2
2 - Installation
Installing the E-MU 1010 PCI Card
Note: Some
E
computer cases don’t use screws to secure PCI cards. In this case, follow the instructions that came with your computer.
PCI Slots
(may not be present
ISA Slots
on your computer)
Align the E-MU 1010 PCI card with the slot and press gently but firmly down into
5.
the slot as shown in figure 2.
Do not force the E-MU 1010 card into the slot. Make sure that the gold finger
6.
connector of the card is aligned with the PCI bus connector on the motherboard before you insert the card into the PCI slot. If it doesn’t fit properly, gently remove it and try again.
Secure the card into the slot using one of the screws you placed aside earlier.
7.
E-MU PCI Digital Audio Systems 13
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2 - Installation Install the Sync Daughter Card or 0202 Daughter Card

Install the Sync Daughter Card or 0202 Daughter Card

E-MU 1616 - If you’re planning to use Word Clock. MIDI Time Code or SMPTE
sync, unwrap the Sync Daughter Card and get ready to install it. If you don’t need these options or don’t have an empty PCI slot, you can skip these next few steps.
• E-MU 0202 - Unwrap the 0202 Daughter Card and get ready to install it.
1.
Connect the ribbon cable provided with the kit between the E-MU 1010 card and the 0202 Daughter card or Sync Daughter Card as shown in figures 3 & 4. The cables are keyed so they cannot be incorrectly inserted. Seat the connectors firmly in the sockets and arrange the cables neatly.
2.
Align the Sync Daughter Card or the 0202 Daughter Card with the slot and press gently but firmly down into the slot as shown in figure 2 on the preceding page.
3.
Do not force the E-MU Card into the slot. Make sure that the tab at the rear of the card is aligned with the PCI bus connector on the motherboard before you insert the card into the PCI slot. If it doesn’t fit properly, gently remove it and try again.
Secure the card into the slot using one of the screws you placed aside earlier.
4.
•
Figure 3 Figure 4
Sync Daughter Card
After all components have been installed and securely fastened, close the computer
5.
0202 Daughter Card
case.
6.
Connect the supplied network-type cable from the 10 BaseT jack on the E-MU 1010 PCI card labeled ā€œEDIā€ to the matching connector labeled ā€œEDIā€ on the MicroDock. The cable supplied with the MicroDock is specially shielded to prevent unwanted RF emissions.
Plug the power cord back into the wall outlet and turn on your computer.
7.

WARNING: E-MU 0202 & MicroDock

If you have both the E-MU 0202 I/O card and the MicroDock, DO NOT connect both to the E-MU 1010 PCI card.

CAUTION: Do not
connect the supplied CAT5 cable to the Ethernet or network connector on your computer. Doing so may result in permanent damage to either your computer, the E-MU 1010 or both.
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Connecting the MicroDock

1.
Connect the supplied EDI cable between the 1010 PCI Card and the MicroDock.
Connect the supplied +48 volt DC adapter to the +48VDC jack on the rear of the
2.
MicroDock. See the diagram below.
3.
Connect your audio inputs and outputs to the MicroDock as shown on page
Turn the MicroDock on by turning the Headphone Volume control.
4.
+48V DC Adapter
48
VDC
+
-
23.
2 - Installation
Connecting the MicroDock
The 1616 and 1616

MicroDocks cannot be used with older 1010 PCI cards identified by the 1394 FireWire port.
m
EDI
The Headphone Volume Control is the Power Switch.
1010 PCI Card
Warning:

The MicroDock has been designed to use readily available and inexpensive standard computer system cables. This makes it easy for you to find replacement cables if your original cable becomes damaged or lost. However, because these standard cables types are used for other purposes, you must use caution to avoid connecting the cables incorrectly. DO NOT connect the supplied EDI cable to the Ethernet or network connector on your computer. Doing so may result in permanent damage to either your computer, the E-MU 1010 PCI card, or the MicroDock.
E-MU PCI Digital Audio Systems 15
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2 - Installation Software Installation

Software Installation

Installing the E-MU 1010 Drivers

The first time you restart your PC after installing the E-MU 1010 PCI card, you will need to install the PatchMix DSP software and E-MU 1010 PCI card drivers.

Windows XP

The software is not compatible with other versions of Windows.
1.
After you have installed your Digital Audio System, turn on your computer. Windows automatically detects the Digital Audio System and searches for device drivers.
2.
When prompted for the audio drivers, click the
3.
Insert the E-MU software Installation CD into your CD-ROM drive. If Windows AutoPlay mode is enabled for your CD-ROM drive, the CD starts running automatiĀ­cally. If not, from your Windows desktop, click (replace and double-click
4. The installation splash screen appears. Follow the instructions on the screen to
d:\
with the drive letter of your CD-ROM drive). You can also open the CD
Setup.exe
.
complete the installation.
5. Choose ā€œContinue Anywayā€ when you encounter the ā€œWindows Logo Testingā€
warning screen. See the note below for more information.
6. When prompted, restart your computer.
Cancel
button.
Start->Run
and type
d:\setup.exe
E
Serial Number -
During the registration process, you will be asked to enter your 18-digit serial number. The serial number is located on the back of the box and on the 1010 PCI Card.

Uninstalling all Audio Drivers and Applications

At times you may need to uninstall or reinstall some or all of the audio card's applica­tions and device drivers to correct problems, change configurations, or upgrade outdated drivers or applications. Before you begin, close all audio card applications. Applications still running during the uninstallation will not be removed.
1. Click Start -> Settings -> Control Panel.
2. Double-click the Add/Remove Programs icon.
3. Click the Install/Uninstall tab (or Change or Remove Programs button).
4. Select the E-MU driver/application entries and then click the Add/Remove (or
Change/Remove) button.
5. In the InstallShield Wizard dialog box, select the Remove option.
6. Click the Yes button. Restart your computer when prompted.
7. You may now re-install existing or updated E-MU 1010 PCI card device drivers or
applications.

Note About Windows Logo Testing

When you install the Digital Audio System drivers, you will see a dialog box that informs you that the driver has not passed Windows Logo testing.
The Digital Audio System drivers are not signed because the driver does not support some of the consumer audio features that the Microsoft driver signing program requires, most notably Digital Rights Management.
However, the Digital Audio System drivers have been rigorously tested using the same test procedures that a signed driver requires, and it passes in all important categories, including those that measure the relative stability of the driver. So, it is perfectly safe to install these drivers on your computer.
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3 - PCI Card & Interfaces

The E-MU 1010 PCI Card

The E-MU 1010 PCI card is the heart of the system and contains E-MU’s powerful E-DSP chip. The powerful hardware DSP on this card leaves more power free on your CPU for additional software plug-ins and other tasks.

Important

When the MicroDock is connected to the 1010 PCI card, the digital I/O on the PCI card is disabled. Use the digital I/O on the MicroDock.

Connections

EDI Connector
Connects to the MicroDock using the supplied EDI cable. This cable provides a a two-way data link
EDI
Connects to
MicroDock
via EDI Cable
S/PDIF In/Out
between the E-MU 1010 and the MicroDock as well as supplying power to the MicroDock.
S/PDIF Digital Audio Input & Output
RCA phono jacks are standard connectors used for S/PDIF (Sony/Philips Digital InterFace) connections. Each jack carries two channels of digital audio. The E-MU 1010 receives digital audio data with word lengths of up to 24-bits. Data is always transmitted at 24-bits.
3 - PCI Card & Interfaces
The E-MU 1010 PCI Card
S/PDIF digital I/O can be used for the reception and/ or transmission of digital data from external digital devices such as a DAT external analog-to-digital
ADAT
or S/PDIF
Optical
In/Out
converter or an external signal processor equipped with digital inputs and outputs.
The S/PDIF out can be configured in either Profes­sional or Consumer mode in the Session Settings menu. The 1010 PCI card can also send and receive AES/EBU digital audio through the use of a cable adapter. See AES/EBU to S/PDIF Cable Adapter for details.
The S/PDIF input and outputs are usable at the
44.1kHz, 48kHz 88.2kHz and 96kHz sample rates, but are disabled for 176.4kHz and 192kHz. The word clock contained in the input data stream can be used as a word clock source. See System Settings.
ADAT Optical Digital Input & Output
The ADAT optical connectors transmit and receive 8 channels of 24-bit audio using the ADAT type 1 & 2 formats. The word clock contained in the input data stream can be used as a word clock source. See System Settings. Optical connections have certain advantages such as immunity to electrical interference and ground loops. Make sure to use high quality glass fiber light pipes for connections longer than 1.5 meters.
 Important: When
using any type of digital I/O such as S/PDIF or ADAT, you MUST sample sync the two devices or clicks and pops in the audio will result.
E-MU PCI Digital Audio Systems 17
Page 18
3 - PCI Card & Interfaces The 0202 Daughter Card
At the 96kHz or 192kHz sample rates, the industry standard S/MUX interleaving scheme is used for ADAT input and output. S/MUX uses additional ADAT channels to achieve the required bandwidth. See the chart below or go here for additional inforĀ­mation.
Sample Rate Number of Audio Channels
44kHz/48kHz 8 channels of 24-bit audio
88.2kHz/96kHz 4 channels of 24-bit audio, using S/MUX standard
176.4kHz/192kHz 2 channels of 24-bit audio, using S/MUX standard

The 0202 Daughter Card

The 0202 Daughter card is the companion card for E-MU 1010 systems which don’t include the MicroDock. The 0202 Daughter card provides one pair of 24-bit balanced analog inputs and one pair of 24-bit balanced analog outputs, plus MIDI in and out.

Connections

Analog Inputs and Outputs
The 0202 Daughter Card provides two balanced, analog inputs and two balanced, line level analog outputs. The inputs can be connected to any line level
Left / Right
Line Inputs
stereo signal from keyboards, CD-players, cassette decks, etc. The analog inputs are assigned to a mixer strip in the mixer application.
The outputs can feed any line level input such as a
Left / Right
Line Outputs
mixing board, the auxiliary input on your stereo or a set of powered speakers. The line outputs are not designed to drive headphones directly. Connect the line outputs to a stereo receiver or mixer with a headphone jack to obtain the proper current drive.
MIDI
In/Out
Either TRS (tip-ring-sleeve) balanced or TS unbalanced cables can be used. Balanced cables provide better noise immunity and +6dB higher signal level. The output line level can be set to accommodate the consumer -10dBV standard, or the pro audio +4 dBu standard in the I/O screen of the Session Settings dialog box. See I/O Settings.
MIDI In/Out
The MIDI input and output port can be assigned in your specific MIDI application. Connect the MIDI adapter cable that came with your 0202 Daughter card to the mini­DIN connectors on the card. The adapter cables convert the mini-DIN to standard DIN connectors used on most keyboards and synthesizers. Connect MIDI Out to the MIDI In port of your synthesizer and MIDI Out of your synth to MIDI In of the 0202 Daughter Card.
18 Creative Professional
Page 19

The MicroDock

The MicroDock connects to the E-MU 1010 PCI card via the EDI cable.
The MicroDock provides (4) balanced analog inputs, (2) microphone preamp inputs, (6) balanced line-level analog outputs, (3) stereo 1/8ā€ outputs for connecting powered computer speakers, (2) MIDI inputs, (2) MIDI outputs, a stereo headphone output, and a RIAA equalized turntable preamp section which is ā€œnormalledā€ into line input 2L and 2R, 8 channels of ADAT digital input/output, and stereo S/PDIF digital input/output.
3 - PCI Card & Interfaces
The MicroDock
f The MicroDock is
completely ā€œhot pluggableā€ā€” It’s OK to plug or unplug the MicroDock while the computer is turned on.
Out
Line
A
In
Mic
Clip
SL
-15
Line -
0
Mic -
1L
1R
1L
1R
Line
B
-3
-6
-12
-20
+50 +65
2L
2L
Mic
Clip
-3
-6
SL
-12
-20
-15
0
Phono
2R
2R
2L
3L
48V
+50 +65
2R
Gnd
3R
S/PDIF
In
MIDI Cable
Out
2
1
The inputs are configured as follows:
(2) mono microphone/line inputs (2 inputs)
(2) stereo pairs of line level inputs (4 inputs)
(1) stereo pair of S/PDIF/AES digital inputs (2 inputs)
(4) stereo pairs of ADAT channels on the ADAT optical input (8 inputs)
Out
 It’s a good idea to
mute MicroDock inputs 2 in the PatchMix DSP
Off
mixer when nothing is plugged in, since the turntable preamp has a very high gain (60dB) and could contribute
48
VDC
+
-
3
EDI
extra noise to your mix/ monitor bus.
(1) RIAA equalized turntable preamp input allows you to connect a turntable without using
an expensive external preamp.
Note: These inputs are automatically disconnected
when plugs are inserted into inputs 2L & 2R.
(2) MIDI input ports using the supplied breakout cable
The outputs are configured as:
(3) stereo pairs of line level outputs
(1) stereo pair driving a stereo headphone jack
(1) stereo pair of S/PDIF/AES digital outputs
(4) stereo pairs of ADAT channels on the ADAT optical output
(3) stereo 1/8ā€ computer speaker outputs. These outputs carry the same signals as the 3
stereo line level outputs and are provided as a convenience for connecting computer or powered speaker systems.
(2) MIDI output ports using the supplied breakout cable
E-MU PCI Digital Audio Systems 19
(Share the same routing as Line Outs 1L/1R)
Page 20
3 - PCI Card & Interfaces The MicroDock

Front Panel Connections

Preamp Section
The front panel mono Mic/Line inputs A & B can be used as balanced microphone inputs, hi-Z guitar pickup inputs, or line level inputs. The Neutrik combination jack accepts microphones using a standard XLR connector or line level/hi-Z inputs (such as an electric guitar) using a standard 1/4 inch TRS/TS connector.
Each preamp has a level control which sets the preamp gain from 0dB to +65dB for the XLR input and from -15dB to +50dB for the Hi-Z line input. The line markings around the knobs are calibrated in 10dB increments. The heavy hash marks on the gain controls indicate unity analog gain to the converter inputs (~5dBV input = 0dBFS output).
A phantom power switch enables +48 volt phantom power supplied to both microĀ­phones. A red LED illuminates to indicate phantom power is enabled. The audio mutes for a second when phantom power is turned on. After turning phantom power off, wait two full minutes before recording to allow the DC bias to drain. See Phantom Power for additional information.
Each microphone input has its own input level meters and clipping indicators. The LED meters indicate signal presence. Adjust the input gain so that the yellow LEDs are illumiĀ­nated. The red Clip LED indicates that the gain is set too high and the signal is clipping the input. These LEDs monitor the signal directly at the analog-to-digital converters and before any processing by the rest of the system. When setting the levels for signals being sent into the MicroDock, the red clip indicator should never flash.
 Phantom Power
Caution: Some microphones (notably ribbon types) cannot tolerate phantom power and may be damaged. Check the specifications and requirements of your microphone before using phantom power.
S/PDIF Digital Audio Input & Output
RCA phono jacks are standard connectors used for coaxial S/PDIF (Sony/Philips Digital InterFace) connections. Each jack carries two channels of digital audio. The MicroDock sends or receives digital audio data at 44.1k, 48k, 88.2k, 96k, 176.4k or 192k sample rates. Data is always transmitted at 24-bits, but lower word widths can be read. The word clock contained in the input data stream can be used as a word clock source. See System
Settings.
S/PDIF digital I/O can be used for the reception and/ or transmission of digital data from external digital devices such as a DAT, external analog-to-digital converter or an external signal processor equipped with digital inputs and outputs.
The S/PDIF out can be configured in either Professional or Consumer mode in the Session Settings menu. The MicroDock can also send and receive AES/EBU digital audio through the use of a cable adapter. See Cables - balanced or unbalanced? for details.
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Page 21
ADAT Optical Digital Input & Output
The ADAT optical connectors transmit and receive 8 channels of 24-bit audio using the ADAT type 1 & 2 formats. The word clock contained in the input data stream can be used as a word clock source. See System Settings. Optical connections have certain advantages such as immunity to electrical interference and ground loops. Make sure to use high quality glass fiber light pipes for connections longer than 1.5 meters.
At the 88.2k, 96k, 176.4k or 192k sample rates, the industry standard S/MUX interĀ­leaving scheme is used for ADAT input and output. S/MUX uses additional ADAT channels to gain additional bandwidth on the existing interface. See the chart below or go here for additional information.
Sample Rate Number of Audio Channels
44kHz/48kHz 8 channels of 24-bit audio
88kHz or 96kHz 4 channels of 24-bit audio, using S/MUX standard interleaving
176kHz or 192kHz 2 channels of 24-bit audio, using S/MUX standard interleaving
The ADAT inputs and outputs can be configured in the System Settings (page 31) to send and receive S./PDIF optical data at 44.1k, 48k, 88.2k, or 96k sample rates.
S/PDIF Optical is not supported at 176.4k or 196k.
3 - PCI Card & Interfaces
The MicroDock
 Important: When
using any type of digital I/O such as S/PDIF or ADAT, you MUST sample sync the two devices or clicks and pops in the audio will result.
Headphone Output & Volume Control
The headphone output drives standard stereo headphones and the adjacent volume control sets the listening level. The headphone amplifier can drive headphones with impedance as low as 24 ohms. The headphone output uses a high-current version of the high-quality output amplifiers used on the other channels. For this reason it has a very clean signal that can be used as another stereo output if you need it.
E Note: PatchMix DSP
does not support AC3 passthrough at this time.
E-MU PCI Digital Audio Systems 21
Page 22
3 - PCI Card & Interfaces The MicroDock
Front Panel Analog Connections
Use the 3-pin XLR jack
for Low Impedance
microphones.
Mic
Line
Line -
Mic -
Mic
Clip
-3
-6
SL
-12
-20
-15
0
A
Use the center
Phone Jack for
High Impedance
Instrument
instruments such
as electric guitar
or bass.
Digital Connections
ADAT
(Optical)
In
Out
Line
B
+50 +65
Mic
Clip
-3
-6
SL
-12
-20
-15
0
+50 +65
48V
S/PDIF
Out
In
Off
On/Off
& Phone Volume
Stereo
Headphones
External A/D - D/A Converter
Out
Optical
1 2 3 4 5 6 7 8
In
(8 more analog inputs & outputs)
DAT or CD
Digital Audio Device with S/PDIF
Out
In
MIDI Keyboard
REAL TIME CONTROLLERS
ASSIGNABLE KEYS
SEQUENCER
Audio Outs
PRESET
SAMPLE
LEVEL
EXIT
ENTER
PAGE
PRESET SELECT
RETURN
EMULATOR
0.987654321
S/PDIF
(Coax)
In
MIDI 1
In
Out
Coaxial
MIDI Out
MIDI In
Out
INC/YES
DEC/NO
ENTER
ESCAPE
Audio Outs
TRIGGERS
ABC
DEF
123
JKL
MNO
GHI
456
TUV
WXY
PRS
789
QZ
0
MIDI
MIDI Sound Module
In
MIDI 2
Out
MIDI In
I
O
VOLUME
22 Creative Professional
MASTER/GLOBAL
TRANSPOSE DIGITAL PROCESSINGSAMPLE MANAGEMENT
MULTIMODE
PRESET MANAGEMENT DYNAMIC PROCESINGPRESET DEFINITION
SAMPLE
PRESET
DRIVE SELECT LOAD SAVE AUDITION TRIGGER MODE
Page 23

Rear Panel Connections

3 - PCI Card & Interfaces
The MicroDock
4 Balanced Line Level Inputs
(configured as 2 stereo pairs)
1L
1R
In
Out
1L
1R
(configured as 3 stereo pairs)
Turntable Inputs
(tied to line input 2)
Phono
2L
2R
2L
2L
2R
2R
3R
3L
Turntable
Ground
Gnd
Out
2
1
Alternate Outputs6 Balanced Line Level Outputs
(same as outputs 1-3)
MIDI Cable
3
MIDI Port
Connector
48 Volt DC
Power Input
48
VDC
+
-
EDI
Connect to
E-MU 1010 PCI Card
Line Level Analog Inputs
4 balanced 24-bit, line-level, analog inputs are provided (1-2). These can be used to input any line level signal from keyboards, CD-players, cassette decks, etc. The analog inputs are assigned to mixer strips in the mixer application. The line level inputs can be set to accommodate the consumer -10dBV standard, or the pro audio +4 dBu standard in the I/O screen of the Session Settings dialog box. See I/O Settings.
The maximum input level is 18dBV (=20.2dBu).
Either TRS balanced or TS unbalanced cables can be used. See page 96 for additional information about unbalanced cables and connectors. The line-level inputs are all servo-balanced, enabling them to convert unbalanced signals to balanced signals internally to reduce noise.
Turntable Inputs & Ground Lug
The RCA turntable inputs feed an RIAA equalized preamp designed for moving magnet type phono cartridges with 60 dB of gain. Connect the ground lead from your turntable to the ground lug to prevent hum.
The turntable inputs share line level inputs 2L and 2R. Inserting a plug into Line Input 2 disconnects the turntable preamp from that channel. Do NOT leave your turntable connected when using inputs 2L and 2R, since this can cause a ground loop.
Important: Do NOT plug in line level signals to the turntable inputs. The turntable inputs are designed to accept the extremely low-level signal from a phonograph cartridge. Use RCA to 1/4ā€ adapters to connect line level signals to the line level analog inputs.
Line Level Analog Outputs
Six balanced 24-bit, line-level, analog outputs are provided (1-3). Output pair 1 is desigĀ­nated as the Monitor Output and is fed by the monitor bus of the PatchMix DSP mixer application. We suggest that you plug your speakers in here. Special anti-pop circuitry mutes the analog outputs when power is turned on or off.
Like the analog line inputs, either TRS balanced or TS unbalanced cables can be used. Balanced cables provide better noise immunity and +6dB higher signal level. The output line level can be set to accommodate the consumer -10dBV standard, or the pro audio +4 dBu standard in the I/O screen of the Session Settings dialog box. See I/O Settings.
The maximum input and output line levels are matched when the input and output settings are set to the same mode (pro or consumer) in the I/O preferences screen.
 It’s also a good idea
to mute the Dock In strip 2L/2R in the PatchMix DSP mixer when nothing is plugged in, since the turntable preamp has a very high gain (60dB) and could contribute extra noise to your mix/ monitor bus.
 Balanced Cables:
You should ONLY use balanced (TRS) cables if BOTH pieces of equipment use balanced connections. Connecting balanced cables between balanced outputs and unbalanced inputs can actually increase noise and introduce hum.
E-MU PCI Digital Audio Systems 23
Page 24
3 - PCI Card & Interfaces
table
The MicroDock
Computer Speaker Analog Outputs
These stereo mini-phone (3.5mm) jacks duplicate line level outputs 1-3 with a lower output level to accommodate consumer speakers. These line level outputs are designed to interface easily with powered speakers.
Computer Speaker Output Duplicates Line Level Output
1 L/R Tip = 1L Ring = 1R
2 L/R Tip = 2L Ring = 2R
3 L/R Tip = 3L Ring = 3R
MIDI 1 & 2 In/Outs
MIDI input and output ports allow you to interface any type of MIDI equipment such as keyboards, effect units, drum or guitar controllers (anything with MIDI). The MIDI drivers were installed when you installed your PatchMix DSP software and the MIDI ports will appear in your system control panel under ā€œSounds and Audio Devicesā€.
There are two completely independent sets of MIDI input and output ports on the MicroDock, which can be assigned in your specific MIDI applications.
Connect the MIDI breakout cable to the D-connector on the MicroDock. Connect MIDI Out to the MIDI In port of your synthesizer and MIDI Out of your synth to MIDI In of the MicroDock MIDI cable.
EDI Connector (Card)
Connects the MicroDock to the E-MU 1010 PCI card using a CAT5-type computer cable. The cable supplied with the MicroDock is specially shielded to prevent unwanted RF emissions.
Basic Connections
Audio
from
Synthesizer
In
Out
Audio
to
Monitors
MIDI Synthesizer
1L
1R
1L
1R
Mixer
Speakers
**
2R
2L
2L
2R
&
MIDI In
Out
MIDI 1
MIDI Out
Phono
2L
2R
3R
3L
Gnd
MIDI Cable
Out
2
1
In
48
VDC
+
3
EDI
Connect
Desktop
Speakers to
1/8" jacks
e
r
o
e
t
S
Turn
-
AC Adapter
1010 PCI
Card
Powered Desktop Speakers
* Note: Line Inputs 2L/2R and Phono 2L/2R cannot be used at the same time.
24 Creative Professional
Page 25
5.1 Surround Speaker Connections
3 - PCI Card & Interfaces
The MicroDock
Center
Left
Front
Phono
2L
2R
1L
In
Out
1L
2L
1R
1R
2L
2R
2R
3R
3L
Gnd
MIDI Cable
Out
2
1
3
Left
Rear
Front Rear Ctr/Sub
Sub-Woofer
(with built-in power amps)
The 1/8ā€ stereo jacks make it easy to connect to powered surround sound speakers. Only three stereo cables are necessary with many speaker systems (see above). The 1/8ā€ jacks duplicate the 1/4ā€ outputs.
48
VDC
+
-
EDI
Right Front
Right
Rear
You can connect the 1/8ā€ stereo jacks to your surround speakers and connect the 1/4ā€ outputs to your other gear for music creation. When you want to monitor in surround, simply open the 5.1 Session and turn on your surround speakers.
The chart below shows how to connect the outputs for 5.1 surround sound playback.
Multichannel WAVE to Surround Sound Speaker Channels
(using the factory 5.1 DVD Playback Session)
WAVE Strip Surround Channels 1/4ā€ Outputs 1/8ā€ Outputs
E-DSP WAVE 1/2 Front Left / Front Right 1L = FL 1R = FR 1 (Tip = FL Ring = FR)
E-DSP WAVE 3/4 Center / Subwoofer 3L = C 3R = Sub 3 (Tip = C Ring = Sub)
E-DSP WAVE 5/6 Rear Left / Rear Right 2L = RL 2R = RR 2 (Tip = RL Ring = RR)
E-DSP WAVE 7/8 Side Left / Side Right N/A N/A
E-MU PCI Digital Audio Systems 25
Page 26
3 - PCI Card & Interfaces The MicroDock
26 Creative Professional
Page 27

4 - The PatchMix DSP Mixer

PatchMix DSP

The PatchMix DSP Mixer is a virtual console which performs all of the functions of a
typical hardware mixer and a multi-point patch bay. With PatchMix, you may not even need a hardware mixer. PatchMix DSP performs many audio operations such as ASIO/
WAVE routing, volume control, stereo panning, equalization, effect processing, effect send/return routing, main mix and monitor control and allows you to store and recall these ā€œSessionsā€ at will.
ī˜‚ To Invoke the PatchMix DSP Mixer
1.
Left-click once on the E-MU icon on the Windows System Tray. The PatchMix DSP mixer window appears.

Overview of the Mixer

Add New
Strip
Physical Input Strips
ASIO Input Strip
Toolbar
4 - The PatchMix DSP Mixer
PatchMix DSP
f Click on the buttons
and knobs in the mixer screen below to jump to the description of the control.
Display
Select
Buttons
Delete
Strip
Channel
Insert
Section
Pan
Controls
Aux
Sends
Volume
Fader
Solo/Mute
Buttons
ā€œTVā€
Screen
Aux
Effects
Section
Sync/
Sample
Rate
Indicators
Monitor
User
Definable
Scribble Strip
E-MU PCI Digital Audio Systems 27
Controls Windows Source Audio
(Direct Sound, Windows Media, etc.)
WAVE Strip
Main
Inserts
Current
Session
Name
Main Mix
Output Volume
& Meters
Volume/Balance
/Mute Controls
Page 28
4 - The PatchMix DSP Mixer Overview of the Mixer

Mixer Window

The Mixer consists of four main sections.
Application Toolbar Lets you manage sessions and show/hide the various views.
Main Section Controls all the main levels, aux buses, and their inserts. This section
also has a ā€œTVā€ which shows parameters for the currently selected effect and the input/output patching. It also shows the session’s current sample rate and whether it’s set to internal or external clock.
Mixer Strips This section is located to the left of the Main Section and shows all
the currently instantiated mixer strips. Mixer strips can represent Physical analog/digital inputs, or Host inputs such as ASIO or Direct Sound. Mixer strips can be added or deleted as necessary. This section can be resized by dragging the left edge of the frame.
Effects Palette This popup window is invoked by pressing the FX button in the
toolbar. Iconic representations of all effects presets are shown here, organized by category. From this window, you can drag and drop effect presets into the insert slots available on the mixer strips and main section aux buses and main inserts.
A simplified diagram of the mixer is shown below.
Input
Post-Fader Strip
Insert
Section
Panning
Input
Pre-Fader Strip
Insert
Section
Fader
MUTE
Aux 1
Aux Bus 1
Aux 1
Send
Amount
Aux
Effects
Insert
Section
Aux 2
Aux Bus 2
Aux 2
Send
Amount
Insert
Section
Fader
MUTE
Main Bus
Return
Amount
Return
Amount

Mixer Block Diagram

Meter
Main Bus
Effects
Insert
Section
Main
Level
Monitor
Out
MUTE
Monitor Level
Main
Out
Output 1L/1R
& Headphones
Pre Fader or Post Fader
When creating a new Mixer Strip, you have the option for the Aux Sends to be placed Post Fader (both Aux Sends come after the channel fader) or Pre Fader (both Aux Sends come before the channel fader). The Pre-fader option allows you to use either Aux Send as another mix bus, which is unaffected by the channel fader. More Information.
28 Creative Professional
Page 29

E-MU Icon in the Windows Taskbar

Right-clicking on the E-MU icon in the Windows taskbar calls the following window.
Right-Click Here
Opens the PatchMix DSP Mixer.
Calls the PatchMix DSP help system.
Disables the splash screen that appears at boot-up.
When unchecked, FX are not loaded until needed, resulting in faster computer boot.
Restores the default PatchMix DSP and driver settings.
Closes the PatchMix DSP background program, disabling use of all audio I/O from the E-MU hardware. Open the PatchĀ­Mix DSP application to start audio again.
4 - The PatchMix DSP Mixer
E-MU Icon in the Windows Taskbar
f Restore Defaults:
Always try this option first if PatchMix is crashing or if you are having any other strange audio problems.

The Toolbar

New
Session
Open
Session
New Session
Open Session Calls up the standard ā€œOpenā€ dialog box, allowing you to open a
Save Session Calls up the standard ā€œSaveā€ or ā€œSave Asā€¦ā€ dialog boxes, allowing
Show/Hide Effects Toggle button that shows or hides the FX palette.
Session Settings Calls up the Sessions Settings window. Session Settings.
Save
Session
ā€œAboutā€
PatchMix DSP
Show/Hide
Effects
Session
Settings
Global
Prefs
Calls up the ā€œNew Sessionā€ dialog box. New Session.
saved Session.
you to save the current Session.
f Click the buttons in
the toolbar to learn about their function.
Global Preferences Calls up the Global Preferences window.
About PatchMix DSP Right-Click on the E-MU logo to view the ā€œAbout PatchMix DSPā€
screen, which provides the software and firmware version numbers and other information.
E-MU PCI Digital Audio System 29
Page 30
4 - The PatchMix DSP Mixer The Session

The Session

The current state of the PatchMix DSP mixer (fader settings, effects routings…everyĀ­thing!) can be saved as a Session. Whenever you create or modify a mixer setup, all you have to do is Save it to be able to recall it at a later time.
Before you begin using PatchMix DSP, you need to set it up to be compatible with the other software applications you may be running. The most important consideration is your system sample rate. PatchMix DSP and any applications or other digital gear you are using must be set to the same sample rate. PatchMix DSP can run at 44.1kHz, 48kHz, 88kHz, 96kHz, 176.4 kHz or 192kHz, but its complete set of features are only available at 44.1kHz or 48kHz. See Chapter 6 - Using High Sample Rates for details.
Once the sample rate is set, you can only easily switch between 44.1k and 48k. You cannot switch between 44/48k and 88k/96k/176k/192k. With a change to these high
sample rates, you must start a new session.
You can also set up an external sync source, thereby obtaining the sample rate from some other device or application. External sync can be obtained from the ADAT input or S/PDIF input. If the session is set at 44.1kHz or 48kHz and the external source is coming in at a higher rate (such as 96k), the Sync Indicator will be extinguished (off), but PatchMix will attempt to receive the external data. The two units are NOT sample locked however, and you should correct this condition to avoid intermittent clicks in the audio. Always check for the presence of the LOCKED indicator whenever you are
using a digital interface.
PatchMix DSP comes with several session templates to choose from so when you create a new session you can either create a ā€œblankā€ session based around a designated sample rate, or select from a list of template starting points.
In a PatchMix DSP session the number of strips in the mixer is dynamically config­urable. This allows you to create only those strips you need up to a maximum number determined by available DSP resources and available inputs.
 Important: When
using any form of digital input, you MUST synchronize the Digital Audio System to the external digital device (S/PDIF/ADAT).

New Session

You create a new session by clicking the ā€œNew Sessionā€ button in the PatchMix DSP main Toolbar. The following dialog box appears.
Select a Template or new Session at the desired sample rate
Session Description
Add your own comment or note about the Session
Check this if you want to edit the New Session.
30 Creative Professional
Page 31
You can now select one of the factory template sessions. The factory templates are pre­programmed with specific setups such as audio recording or mixing. The selector tabs categorize Template Sessions into three groups based on sample rate, 44.1k/48k, 88k/ 96k, or 176k/192k.
You can create your own templates by simply copying or saving sessions into the ā€œSession Templatesā€ folder (Program Files\Creative Professional\E-MU PatchMix DSP\Session Templates).
There is also a Comment area that you can use to give yourself some clue as to what you were thinking when you created the session.
Selecting a Session at 176.4kHZ or 192kHz
When operating at 176.4k or 192k sample rates, the number of I/O channels are slightly reduced. At these high sample rates you must select one of three types of sessions each containing a different I/O configuration. Please see page
87 for details.

Open Session

To Open a saved session, click on the Open Session button. A dialog box appears allowing you to choose one of your saved Sessions to open. Choose one of your saved sessions and click on the Open button.
4 - The PatchMix DSP Mixer
The Session

Save Session

To Save a session, click on the Save Session button. A Save dialog box appears allowing you to choose a location in which to save the current Session. The ā€œMy Sessionsā€ folder is chosen by default.
Get in the habit of saving the session whenever you have created a special mixer setup. This will make your life much easier as you can recall a setup for many different audio modes such as: recording, mixing, special ASIO routings, etc.

Session Settings

System Settings
Pressing the Session Settings button on the toolbar brings up the System Settings window shown below. Click the tabs to select System or I/O options.
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4 - The PatchMix DSP Mixer The Session
The System Settings include the following:
• Internal/External Clock Selects between internal or external word clock source
as the master clock source for the system
• Sample Rate Selects the sample rate when using internal clock. Your choices are: 44.1kHz, 48kHz, 88.2kHz, 96kHz,
176.4kHz, 192kHz.
• External Clock Source
(ext. clock only)
Select from: ADAT, or S/PDIF as an external sample clock source.
Using External Clock
Whenever you are using any digital I/O such as ADAT or S/PDIF, one of the digital devices MUST supply the master clock to the others. This master clock runs at the system sample rate and can be embedded into a data stream such as S/PDIF or ADAT. Common symptoms of unsynced digital audio include, random clicks or pops in the audio or failure of the digital stream to be recognized. Always check for the presence of
the ā€œLOCKEDā€ indicator whenever you are using a digital interface.
If an External Clock is interrupted or switched after the Session has been created (except between 44.1k <-> 48k), the ā€œLOCKEDā€ indicator will be extinguished and PatchMix will attempt to receive the external data. The two units are NOT sample locked however, and you should correct this condition to avoid intermittent clicks in the audio.
I/O Settings
You can set the level (-10dBV or +4 dBu) for each pair of analog outputs and the input gain setting for each pair of analog inputs.
An output setting of +4 provides the most output and is compatible with professional audio gear. Balanced output cables provide a +6dB hotter signal than unbalanced cables when used with balanced inputs. Do NOT use balanced cables unless your other gear has balanced inputs. See ā€œCables - balanced or unbalanced?ā€ in the Appendix for more information.
E Note: if set to
ā€œExternalā€ without an external clock present, PatchMix DSP defaults to the internal 48kHz clock rate.
Comparison of -10dBV & +4dBu Signal Levels
Consumer
Clipping -->
Headroom
+ 6 dBV
+ 2 dBV
{
-10 dBV
0 dBV = 1V RMS 0dBu = .777V RMS
An input setting of -10 is compatible with consumer audio gear and works best with low level signals. (-10dBV is approximately 12dB lower than +4dBu.) Choose the setting that allows you to send or receive a full scale signal without clipping.
Setting correct input and output levels is important! You can measure the level of an input by inserting a meter into the first effect location in the strip. Adjust your external equipment outputs for the optimum signal level. See ā€œTo Set the Input Levels of a Stripā€ for details.
Professional
(balanced)(unbalanced)
+20 dBu
=
+8 dBu
=
+4 dBu
=
-8 dBu
<-- Clipping
Headroom
}
f Input too weak?
Use -10 Input setting.
Output too weak?
Use +4 Output setting
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4 - The PatchMix DSP Mixer
The Session
Input Level
Settings
Optical
Input
Select
Mic Soft Limiting
On/Off
• Inputs +4 or -10 Selects between Consumer level (-10dBV) or
Professional level (+4dBu) inputs. (Use the -10dBV setting if your input is too weak.)
• Outputs +4 or -10 Selects between Consumer level (-10dBV) or
Professional level (+4dBu) outputs. (The +4 dBu setting outputs a hotter level.)
• Optical Input Select Selects between ADAT or optical S/PDIF for the MicroDock
ADAT Input. The coaxial S/PDIF input is disabled when S/PDIF optical is selected.
• Microphone Input
Soft Limiting
• Optical Output Select Selects between ADAT or optical S/PDIF for the MicroDock
• S/PDIF Output Format Selects between S/PDIF or AES/EBU format for S/PDIF. This
The Mic/Hi-Z inputs have built-in ā€œsoft limitersā€ which automatically turn down the gain before the signal overloads the A/D converters. The soft limiters allow you to record a hotter signal without fear of clipping. This control turns the soft limiters On or Off. See
the Best Possible Recording for additional information
about the soft limiters.
ADAT Output. The coaxial S/PDIF Output is disabled when S/PDIF optical is selected.
sets the S/PDIF-AES status bit, but does not affect the signal level.
Output Level Settings
Optical Output
Select
S/PDIF Output Format
Making
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f Physical input strips
are shown with BLUE text.
f Host input strips are
shown with WHITE text.
f The Input Type will
turn RED if the input is not available. (The MicroDock may be disconnected.)

Input Mixer Strips

Input Mixer Strips
PatchMix DSP Input Mixer Strips are stereo except for the MicroDock Mic/Line inputs. Each input mixer strip can be divided into four basic sections.
• Insert Section Effects, EQ, External/Host Sends & Returns can be inserted into the signal path.
• Pan Controls These controls position the signal in the stereo sound field.
• Aux Sends Used to send the signal to sidechain effects or to create separate mixes.
• Volume Control Controls the output level of the channel.
Mono/Stereo

Input Type

Insert Section

Pan Controls

Aux Sends

Channel

Volume Control

Input Type
The very top of the strip is labeled mono or stereo and displays the type of the assigned input. Input mixer strips can be added as desired and can be configured to input the following:
• Physical input = Hardware
(Analog/SPDIF/ADAT).
• Host Input = Software
(Direct Sound, WAV, ASIO source)

Inserts

You can drag and drop effects from the Effects Palette or Right-click to insert a Physical or ASIO Send or Send/Return A Peak Meter, Trim Control or Test Signal can also be inserted by RightĀ­clicking on the Insert section.
Pan Controls
These controls allow to you position the channel in the stereo sound field. Dual controls on stereo strips allow you to position each side indepen­dently.
Aux Sends
These controls send the signal to sidechain effect processors such as reverb and delay. They can also be used to create separate mixes for the artist or for recording.
This screen shows a mono strip on the left and a stereo strip on the right.
34 Creative Professional
Mute/Solo
Buttons
Scribble Strip
Vol ume Control
Controls the output level of the strip into the main/monitor mix bus.

Mute/Solo Buttons

These convenient buttons allow you to solo or mute selected channels.

Scribble Strips

Click inside the scribble strip and type a name of up to eight characters.
Page 35

Mixer Strip Creation

f CDs & MP3s: The
WAVE 1/2 strip is used to playback CDs, Windows Media Player, and Direct Sound.
PatchMix DSP is a dynamically configurable mixer. Each mixer session can contain an arbitrary number of strips up to a limit set by the number of available input sources and available DSP resources.
• Host refers to a computer application such as Cubase.
• Physical refers to a hardware input or output such as an output jack.
ī˜‚ To Add a New Strip:
1. Click on the New Mixer Strip button. See Overview of the Mixer. The New Mixer
Strip Input Dialog appears:
4 - The PatchMix DSP Mixer
Mixer Strip Creation
f Adding or deleting a
strip ā€œdefragmentsā€ the effect/DSP resources. If any effect you wish to add is unavailable (greyed-out), try deleting an unused strip to free up resources.
2. Select the desired input to the mixer strip from the following choices:
• Physical Source: Analog or digital input (Analog, ADAT, S/PDIF)
• Host - ASIO Source input Streaming audio from an ASIO software application.
• Host - WAVE input Window sound sources — WAVE, WDM, CD
PHYSICAL SOURCE Function
Physical: Dock Mic/Line 24-bit monophonic analog input on the MicroDock.
Physical: Dock In 24-bit stereo analog input on the MicroDock.
Physical: Dock S/PDIF 2 channel digital audio from the S/PDIF input on the
Physical: Dock ADAT 2 channel (x4 strips) digital audio from the ADAT input on
HOST SOURCE Function
Host ASIO Output Source
From software application
Host Windows Source
From Windows
3. Select Pre-Fader Aux Sends or leave the box unchecked for Post-Fader Aux Sends.
4. Click OK to create a new strip or Cancel to cancel the operation.
E-MU PCI Digital Audio System 35
MicroDock.
the MicroDock.
2 channel digital audio from an ASIO source (software app). ASIO: 1/2, 3/4, 5/6, 7/8, 9/10, 11/12, 13/14, 15/16
Direct Sound, WDM, Windows Media
(Sound generated or handled by Windows.)
WAVE 1/2 - Default stereo source such as game sound, CD player, beep sounds, etc. WAVE 3/4, WAVE 5/6, WAVE 7/8 - Additional WDM channels
f See ā€œPre or Post Fader
Aux Sendsā€ on page 46.
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4 - The PatchMix DSP Mixer Mixer Strip Creation
ī˜‚ To Delete a Mixer Strip:
1. Click the top of the mixer strip you wish to delete. A red border appears around
the strip, indicating that it is selected.
2. Click on the Delete Mixer Strip button, or right-click and choose Delete, or use the
Delete key on the PC keyboard. See Overview of the Mixer.

Multichannel WAVE Files

The 1616 supports 2 channels of WAVE recording and 8 channels of multichannel WAVE playback. The WAVE channels are available for the following types of WDM
devices:
• Classic MME
• DirectSound
• Direct WDM / Kernel Streaming (KS)
DirectSound and the WDM/KS interfaces allow up to Eight channels of Wave Out while the classic MME interface only exposes 2 channels.
The WAVE channels operate at all sample rates. For additional information about WDM behavior at high sample rates, see page 87.
192kHz/96kHz DVD-Audio disks are protected against digital copying. Most DVDĀ­Audio disks contain duplicate 48kHz audio tracks which will play back on the 1616.
Windows Media Player/DVD/Surround Sound Playback
Select DirectSound as the output format when using Windows Media Player and other DVD player applications.
Eight channel WAVE playback supports either 5.1, 6.1 or 7.1 surround audio. However, the 1616 is best suited to play 5.1 surround, since it only has 6 analog outputs. (You could play back 7.1 surround audio by using an external S/PDIF to Analog Converter. Create a 7/8 WAVE strip and insert a Send to S/PDIF Out.)
The chart below shows how to connect the outputs for 5.1 surround sound playback.
Multichannel WAVE to Surround Sound Speaker Channels
(using the factory 5.1 DVD Playback Session)
WAVE Strip Surround Channels 1/4ā€ Outputs 1/8ā€ Outputs
E-DSP WAVE 1/2 Front Left / Front Right 1L = FL 1R = FR 1 (Tip = FL Ring = FR)
E-DSP WAVE 3/4 Center / Subwoofer 3L = C 3R = Sub 3 (Tip = C Ring = Sub)
E-DSP WAVE 5/6 Rear Left / Rear Right 2L = RL 2R = RR 2 (Tip = RL Ring = RR)
E-DSP WAVE 7/8 Side Left / Side Right N/A N/A
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Page 37

Insert Section

The Insert Section is next in line. PatchMix DSP effects can be selected from the Effects Palette and dropped into the insert locations. See ā€œThe Effects Paletteā€. Any number of effects can be inserted in series.
The Inserts also have the unique ability to patch into ASIO/WAVE and external equipment. ASIO/WAVE Sends, External Sends and External Send/Returns can be dropped into the insert section to route the signal anywhere you want.
The Insert/Patch Bay is incredibly flexible. Want to send the input of the strip to your audio recorder? Simply insert an ASIO send into the insert section and select the ASIO pair you want. That’s it! That input is now available in your ASIO software.
The following types of inserts can be selected.
Hardware Effect Reverb, EQ, Compressor, Flanger, etc. using PatchMix DSP’s effects
which do not load your CPU.
Host ASIO Send Splits off the signal and sends it to an ASIO host input such as a
software audio recorder or anything that uses ASIO.
ASIO Direct Monitor
Ext. Send/Return Sends signal to a selected external output, then returns it to the chain
External Send Sends the signal to an external output. See ā€œTo Add a Send Insert:ā€
Peak Meter Peak meters allow you to monitor the signal level anywhere in the
Trim Pot You can insert a gain control with up to 30 dB of gain or attenuation.
Test Tone This special insert outputs a calibrated sine wave or noise source,
Sends the signal to a selected ASIO host input, then returns a selected ASIO host output to the chain.
via a physical input.
.
chain. See ā€œ
A peak level meter and phase inverter are also included.
See ā€œ
which can be used to track down audio problems.
See ā€œ
Meter Insertsā€.
Trim Pot Insertā€.
Test Tone/Signal Generator Insertā€.
4 - The PatchMix DSP Mixer
Mixer Strip Creation
f You have to create an
ASIO strip or ASIO Send in order to activate these ASIO channels in your software.
Working with Inserts
The Inserts are one of most powerful features of the PatchMix DSP system as they allow you to configure the mixer for a wide variety of applications.
ī˜‚ To Add an Effect to an Insert Location:
1. Press the FX button. The effects palette appears.
2. The effects are organized into categories. Click on a folder to open it.
3. Select the effect you want, drag it over the insert section, then drop it into an insert
location.
4. To rearrange the order of effects, simply drag and drop them into the desired order.
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4 - The PatchMix DSP Mixer
r
Mixer Strip Creation
The Insert Menu
Right-Clicking over the insert section brings up a pop-up selection box containing various insert options to help you control and manage your inserts.
ī˜‚ To Add a Send Insert:
This type of insert send splits the signal at the insert point and sends it out to the selected destination. (An ā€œASIO Sendā€ becomes an input on your recording application, a ā€œPhysical Outā€ goes to a pair of output jacks. the signal also continues down the strip to the Aux Sends and main mixer outputs.)
1. Right-Click over the Insert section. A pop-up dialog box appears.
2. Select Insert Send (to ASIO/WAVE or physical output) from the list of options. The
following dialog box appears.
Input
Insert
Send
Panning
Fader
Aux 1 Bus
Aux 2 Bus
Main Output Bus
3. Choose one of the Send Outputs. Click on a destination to select it.
4. Click OK to select the output or Cancel to cancel the operation.
To ASIO, WAV o Physical Output
ī˜‚ To Add a Send/Return Insert:
This type of insert send breaks the signal at the insert point and sends it out to the selected destination such as an external effect processor. A return source signal is also selected which returns the signal to the channel strip after processing.
1. Right-Click over the Insert section. A pop-up dialog box appears.
2. Select ā€œInsert Send/Return (Physical Output and Input)ā€ from the list of options.
The following dialog box appears.
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t
Mixer Strip Creation
Input
Insert
Send/Return
Panning
Fader
Aux 1 Bus
Aux 2 Bus
Main Output Bus
To Physical Output From Physical Inpu
3. Choose one of the Send Outputs. Click on a destination to select it.
4. Choose one of the Return Inputs. Click on a source to select it.
5. Click OK to select the Send and Return, or Cancel to cancel the operation.
ASIO Direct Monitor Send/Return
This type of insert send breaks the signal at the insert point and sends it out to the selected ASIO Host Input destination (such as Cubase or Sonar). A return source signal is also selected which returns the signal to the channel strip from an ASIO Host Output.
The ASIO Direct Monitor Send/Return is unique in that it utilizes ASIO 2.0 zero-latency monitoring. In order to utilize this feature, Direct Monitoring must be enabled in the
audio recording application.
While recording, the Direct Monitor Send/Return routes the signal to the recording application, but monitors directly from the input to eliminate latency. During playback, the recording application automatically switches the Direct Monitor Send/Return to monitor the recorded track.
 If the source or
destination you want to use is not available in the list, they are probably already being used elsewhere. Check the input Strips, Inserts and Output Assignments.
Input Input
Recording
Software
Direct MonDirect Mon
Recording
Software
Recording Playback
The Direct Monitor Send/Return also allows the recording application to control volume and pan. Normally when using direct monitor recording you’ll want to control the volume and pan from the recording application. In this case, set the PatchMix DSP stereo pan controls hard left and right, mono pan controls to center, and the fader to 0dB.
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ī˜‚ To Add an ASIO Direct Monitor Send/Return:
1. Right-Click over the Insert section. A pop-up dialog box appears.
2. Select Insert ASIO Direct Monitor from the list of options. The following dialog
box appears.
3. Choose one of the Send Outputs. Click on a destination to select it.
4. Choose one of the Return Inputs. Click on a source to select it.
5. Click OK to select the Send and Return, or Cancel to cancel the operation.
Meter Inserts
Keeping track of signal levels is important in any audio system, be it analog or digital. You want to keep the signal levels running as close to maximum in order to achieve high resolution and low noise. On the other hand, you don’t want the signal level so high as to cause clipping. To help you maintain optimum signal levels, we have included Peak Level Meters, which can be dropped into any insert location.
The insert meters are of the ā€œpeak holdā€ type. The topmost bar in the meter holds its highest level for a second to let you see transients that would otherwise be too quick for the eye.
The peak meters are also color-coded to indicate the signal strength. The chart below outlines the meanings of the colors. Avoid lighting the topmost red bar, as this indicates distortion of the signal. Click on the clip indicator to turn it off.
Meter Color Indicates
E Red Indicates signal clipping.
E Yellow Good strong signal level.
E Green Signal is present.
One of the most obvious uses of the insert meters is to set input levels. On the analog inputs, the analog-to-digital converter (ADC) is one of the most critical points in the signal path. You want the input signal level to drive the 24-bit ADCs into their optimum range without clipping. A reading of 0dB on an input meter indicates signal clipping.
Level
10203040506070
--12dB
Each bar of the meter equals 1dB. The yellow bars begin at -12dB below full scale.
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The insert meters are also useful to monitor incoming digital signals such as ADAT, ASIO or S/PDIF to make sure the mixer is receiving a proper signal level. They’re also great for troubleshooting, since you can place them virtually anywhere in the mixer.
ī˜‚ To Insert a Meter
1. Right-Click on an Insert location of the mixer strip. A pop-up dialog box appears.
2. Select Insert Peak Meter. A stereo peak meter appears in the insert location.
3. Select Effect in the Main Section. The meters are now shown in high resolution in
the TV screen.

To Set the Input Levels of a Strip

1. Select the topmost Insert location on a mixer strip and insert a meter (see above).
2.
Left-click on the meter insert to see the meter in the TV screen.
3. Feed your audio signal to the input of the mixer strip. The meter should now show
the signal level.
4. Adjust the output level of the external device (synthesizer, instrument, preamp,
etc.) feeding the MicroDock. The meter should be in the yellow region most of the time with occasional forays into the red. If the clip indicator ever comes on, reduce the signal level.
5. Each analog input pair has its own Input Pad (-10dBV or +4dBu) which controls the
input signal range. Changing the I/O settings can add or subtract 12dB. Check these settings if you cannot set the proper input level. See I/O Settings.
4 - The PatchMix DSP Mixer
Mixer Strip Creation
f Input too weak?
Use -10 Input setting.
Output too weak?
Use +4 Output setting
Making the Best Possible Recording
Making a good digital recording is easier than ever thanks to the high resolution 24-bit A-D converters on your Digital Audio System. These converters are much more forgiving than the 12-bit or 16-bit converters of the past. Even so, to get the best performance possible, you'll need to follow a few basic guidelines.
First, whenever you input an analog signal to the Digital Audio System, make sure that you're feeding the A-D converters with an optimum signal level. The quality of a digital recording is directly related to the signal level you feed into the A-D converters. If the analog input level is set too low, you lose resolution—if it's set too high, the A-D converters will clip.
To measure the input level, simply add an insert meter to the channel strip in PatchMix DSP. These meters are accurately calibrated to display 1dB for each bar on the meter. You can enlarge the meter view by clicking on the insert meter in a strip and selecting the ā€œEffectā€ button at the top of the TV screen.
The ā€œI/O Settingsā€ in the Digital Audio System allow you to set the input levels to
-10dBV (consumer equipment level) or +4dBu (professional equipment level) for each analog input. This control sets the overall input level to match your other gear, but to get the best possible recording you need to fine tune the level further.
In order to supply the correct input level, you’ll need to adjust the output of your analog source (electric instrument or preamp) so that the input level comes close to 0dB without ever going over.
Play your input source signal while watching the insert meter in the strip. The signal should go into the yellow area frequently, but never into the red. Adjust the level of your source until you have a good level. If the signal is way too strong or too weak, you may
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have to go back and adjust the I/O Settings. Choose ā€œ-10ā€ if the input signal is too weak and ā€œ+4ā€ if the signal is too strong.
Digital audio has NO headroom past 0dBFS (FS = Full Scale) and will ā€œhard clipā€ if the signal exceeds 0dBFS. Hard clipping sounds bad and will ruin your recording. Hard clipping occurs because at 0dBFS, all 24 bits are turned on and the A-D cannot measure any higher level. Analog tape, unlike digital, can be driven past 0dB, although with some degradation of the signal.
The MicroDock includes a pair of Soft Limiters on the preamp inputs, which can be turned on or off for each channel in the I/O Settings. The soft limiters automatically turn down the gain whenever the signal level exceeds -6dB below Full Scale. Below this level, the limiters are completely out of the circuit. The soft limiters allow you to encode a hotter signal without fear of hard clipping the input. This provides increased resolution and a better recording. When recording drums, piano and vocals, occasional peak transients can be tamed by the soft limiters, allowing you to supply the best possible signal into the MicroDock’s ultra-high-quality A-D converters.
The Digital Audio System includes Insert ā€œTrim Potā€ controls, but since they adjust the signal level AFTER the signal has been digitized, this will not recover any lost resolution. It’s far better to set the input level correctly in the first place. Trim Pots can be used in emergency situations if there's no other way to get a hot signal in. They are designed to optimize the signal levels feeding effect plug-ins.
Trim Pot Insert
The Trim Pot Insert allows you to adjust the level of a signal in an insert location. The trim pot provides up to ±30dB of gain or attenuation and a phase inverter. The trim pot also has a built-in stereo peak meter after the control.
Gain/Attenuation
Phase Invert
Meters
You might use a trim pot to boost or attenuate a send or return from an external effect, or to drive an effect device. Certain effects such as the Compressor, Distortion, and Auto-Wah are very level dependent and like to see a good, strong input signal. If you are working with a weak signal, you can improve the performance of these effects inserting a trim pot and boosting the gain.
Trim pots can be used to boost the level of analog line level inputs in a pinch, but it’s much better to boost the signal level before the A/D converters in order to get maximum resolution and signal-to-noise ratio from the converters.
The phase invert switch inverts the polarity of the signal. It is generally used to correct for balanced lines and mics that are wired backwards.
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ī˜‚ To Add a Trim Pot Insert
1. Right-Click over any of the Insert sections. A pop-up dialog box appears.
2. Select Insert Trim Control from the list of options. A Trim Pot insert appears in the
insert location.
3. Click on the Trim Pot insert to view and adjust the controls in the TV screen.
4. To move the Trim Pot to another location, simply drag and drop it into the desired
position.
Test Tone/Signal Generator Insert
The test tone/signal generator insert is a handy troubleshooting aid which outputs a calibrated sine wave, white noise or pink noise. This tool, in combination with an insert meter, allows you to accurately measure the signal gain or attenuation of an internal or external device. The test tone can also be quite handy for tuning up musical instruĀ­ments.
Signal Type
(Sine wave, White or Pink Noise)
4 - The PatchMix DSP Mixer
Mixer Strip Creation
f Musical Note Freq.
A = 440 Hz B = 493.88 Hz C = 523.25 Hz D = 587.33 Hz E = 659.26 Hz F = 698.46 Hz G = 783.99 Hz
Sine Wave Oscillator Frequency
Test Signal Output Level
The Sine Wave Oscillator frequency is variable from 20Hz-20kHz. The level is variable from off to +30dB.
White Noise is a mixture of all frequencies in the audio spectrum at the same average level (analogous to white light in the visible spectrum).
Pink Noise provides equal power distribution per octave. (White noise has more power in the higher octaves.) Pink noise and white noise are useful as wideband sound sources.
ī˜‚ Using the Test Tone and Meter Inserts for Troubleshooting
Sometimes it’s useful to have a continuous tone to verify that you have the signal path routed correctly in hardware or software. First insert a Test Tone and/or a Meter(s) into a strip, then follow the tone through the system by ear or by moving the meter. A test tone is quite handy when first setting up your recording software.
1. Right-Click over the Insert section in question. A pop-up dialog box appears.
2. Select Insert Test Tone/Signal Generator from the list of options. A Test Tone insert
appears in the insert location.
3. Click on the Test Tone insert to view and adjust the controls in the TV screen.
4. To move the Test Tone to another location, simply drag and drop it into the
desired position.
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4 - The PatchMix DSP Mixer Mixer Strip Creation

Managing Your Inserts

ī˜‚ To Delete an Insert:
1.
Right-Click over the Insert you wish to delete. A yellow line around the insert location indicates that it is selected. A pop-up dialog box appears.
2. Select Delete Insert to remove the selected insert or select Delete All Inserts to
remove all inserts.
3. The insert(s) are deleted from the insert chain.
ī˜‚ To Bypass an Insert:
Inserts can be bypassed if you want to temporarily hear the audio without the effect or insert. Bypass can also be used to turn off a Send Insert.
Method #1
1. Click on the Effect (in the Insert section) and select Effect in the TV display.
2. Click the Bypass button.
Method #2
1. Right-Click over the Effect you want to bypass (in the Insert section). A pop-up
dialog box appears.
2. Select Bypass Insert from the list of options.
f Tip: Select the Insert
and press the Delete key to delete the plug-in from the strip.
ī˜‚ To Bypass All Inserts:
All Inserts in a strip can also be bypassed with a single command.
1. Right-Click over the Effect you want to bypass (in the Insert section). A pop-up
dialog box appears.
2. Select Bypass All Inserts from the list of options.
ī˜‚ To Solo an Insert:
Inserts can also be soloed. Solo bypasses all the other inserts in the strip and allows you to hear only the soloed effect. This feature is very useful when adjusting the effect parameters.
Method #1
1. Click on the Effect (in the Insert section) and select Effect in the TV display.
2. Click the Solo button.
Method #2
1. Right-Click over the Effect you want to Solo (in the Insert section). A pop-up dialog
box appears.
2. Select Solo Insert from the list of options.
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Page 45

Aux Section

The Auxiliary Sends tap the signal from the channel strips and sum them together before sending the mix to the Auxiliary Effects section. In a traditional mixing console, aux sends are used to send part of the signal to outboard effect devices, then return the effected signal back into the mix using the effect returns. This is called a Sidechain Routing because the aux signal takes a detour through the effects before being summed back into the main mix. Sidechain effects are usually effects that you might want applied to several channels, such as reverb.
Incidentally, the wet/dry mix of effects in the Aux Sends should normally be set to 100% wet. This is because you will be adjusting the effect amount using the Aux Return control instead. If you have more than one effect in an Aux Bus, ignore the preceding advice as the wet/dry controls can be used to mix the amounts of your multiple effects.
The Aux 1 & 2 buses can also be used as additional submix output buses just like the main output. Simply drop an ASIO or External Send Insert into the chain and the stereo bus is sent. Turn off the Return Amount if you don’t want the submix to be combined into the main mix.
Aux Send and Return values can also be changed by typing directly into the displays.
4 - The PatchMix DSP Mixer
Mixer Strip Creation
Input
Pan
Fader
Mute
Amt
Amt
Sidechain Diagram
(Post-Fader Aux Sends)
Send
Amount
Aux Bus 1
Chain
Send
Amount
Aux Bus 2
Chain
Main / Monitor Bus
Return
Amount
Side
Return
Amount
Side
Output
Other Uses of the Aux Sends
You can think of the Aux Sends as two extra mixing buses because that’s exactly what they are. These two mixes can be routed anywhere, such as to a physical output or an ASIO pair. You could route one of the Aux buses to the Monitor out to create a monitor mix while sending the main mix off to your audio recording software.
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4 - The PatchMix DSP Mixer Mixer Strip Creation
Pre or Post Fader Aux Sends
When you create a New Mixer Strip you have the option to place both Aux Sends after the channel volume fader and mute control or you can place them before the fader and mute. Post-Fader turns down the send level as you lower the volume of the strip. With Pre-Fader selected, you may still hear the effected signal returning from one of the Aux Buses with the volume fader turned down.
With the Pre-Fader box selected, the Aux Send levels are completely unaffected by the Level Fader and Mute settings. The Pre-Fader setting allows you to create two completely different mixes using the Aux Buses since the signal levels of this mix won’t be affected by the fader settings.
Input
Volume Fader & Mute does NOT affect Send Levels
Pre-Fader Aux Send
Pan
Return
Amount
Return
Amount
Amt
Amt
Aux Bus 1
Send
Amount
Side
Chain
Send
Amount
Side
Fader
Aux Bus 2
Chain
 In order to change a
strip from pre-fader to post-fader or vice-versa, you have to delete the strip and create a new one.
Pan
Fader
Mute
Mute
Input
Amt
Amt
Main / Monitor Bus
Post-Fader Aux Send
Volume Fader & Mute affects both Aux Send Levels
Send
Amount
Aux Bus 1
Send
Amount
Aux Bus 2
Output
Return
Amount
Side
Chain
Return
Amount
Side
Chain
Main / Monitor Bus
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Output
Page 47

Level, Pan, Solo & Mute Controls

The Pan control comes before the Level Control
Pan Controls
Aux Send
Amount
Controls
Level Control
Mute & Solo
Buttons
Scribble Strip
and Aux Sends in the signal flow. On stereo strips we use an unconventional pan section with two pan pots – one for the left part of the signal and one for the right part of the signal. This feature allows you to independently position both sides of the stereo signal. A conventional stereo balance control only allows you to turn down one side or the other.
The Mute button does just what you would expect—press the button and the sound from that channel is cut off. Pressing the Solo button while the Mute button is pressed allows you to hear the channel until solo is turned off.
The Solo button allows you to listen to only that channel while muting the rest of the inputs. If multiple solo buttons are pressed, you will hear all soloed channels and the non-soloed channels will all be muted.
The mute status is remembered if a muted channel is soloed. When the channel solo is turned off, the channel reverts to being muted.
The Level Control for the strip is an attenuation control that can also provide up to +12dB of gain. 0db is the unity gain setting. You can also type numeric values into the displays to set the level.
At the very bottom is the Scribble Strip text area, into which you can type any short piece of text, thus naming the strip, i.e. ā€œvocalsā€, ā€œbassā€, ā€œdrumsā€ and so on.
4 - The PatchMix DSP Mixer
Mixer Strip Creation
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4 - The PatchMix DSP Mixer Main Section

Main Section

Physical/Host
Select Buttons
ā€œTVā€ Screen
View
Selection
Buttons
Aux
Insert
Section
Master
Aux Send
Amounts
Main
Insert
Section
Output
Fader &
Meters
Master Aux
Return
Amounts
Sync &
Sample Rate
Indicators
Monitor Controls
Session Name
The main section contains all controls for controlling the main mix elements as well as a ā€œTV screenā€ for viewing the input/output routing or parameters of the selected insert.
The three buttons across the top of the main section select what is shown on the TV display. Input and output routings are graphically displayed. When an insert is selected (by clicking on the insert), the screen shows the available parameters for the currently selected insert.
Below the TV screen is the Aux Bus section where effects, effects chains or other inserts can be assigned to the two aux buses. Send and return levels can be individually controlled for each of the two Aux Buses.
The Aux 1 and Aux 2 buses are fed by the two Aux Sends on each mixer strip. The Master Send Level control on Aux bus 1 and 2 can be used to attenuate or boost the signal going into the Auxiliary Inserts. There is also a Master Return Level to control the amount of the effected signal that will be returned into the main mix.
The Main Bus can also have a chain of effects inserted. (You might put an EQ here to equalize your entire mix or add an ASIO or WAVE send to record the mix.) Note that the Main Output level control comes before the Monitor Level so that you can control the monitor level without affecting the level of your recording mix or main mix. There is a stereo peak meter that indicates the signal strength for the main mix.
The Monitor section has a volume, balance, and a mute control to cut off the monitor output.
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TV Screen & Selectors

The ā€œTV screenā€ at the top of the main section is a multi-function display and control center for the input and output routings and effect controls. The three buttons at the top of the display select the current function of the display—Effect, Inputs or Outputs.
Effect
Select the Effect display view in the main section, then click on an Effect Insert to display the effect parameters. If an insert effect is not selected, the display will read ā€œNo Insertā€.
Most effects have a wet/dry mix parameter to control the ratio of effect to plain signal. The wet/dry setting is stored with the effect preset. The parameter set varies with the type of effect. See ā€œList of Core Effectsā€ for detailed information about the individual effects.
4 - The PatchMix DSP Mixer
Main Section
E Note: Effects have to
be placed into an insert location before you can program them.
Effect Display
View Button
Wet/Dry Mix Control
Effect Location
Effect Bypass & Solo Buttons
Effect Parameters
User Preset Section
When a Send or a Send/Return insert is selected with the effects display enabled, the TV screen shows you where the Send is going and where the Return is coming from. The bypass or solo buttons at the top of the display are available for Send/Return type inserts only.
Send Destination
Return Source
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4 - The PatchMix DSP Mixer Main Section
Input
Selecting the Input display view shows a graphic representation of the PatchMix DSP Mixer inputs. This screen is only a display, unlike the Effects and Outputs screens, which allow you to make routing changes. Input routing changes are made by adding mixer strips. See Mixer Strip Creation.
The input routings are divided into two categories: Physical Inputs and Host Inputs. Select either category by clicking on the Physical or Host button.
Physical Input Display Host Input Display
Return
Amount
f The Input and Output
displays make it much easier to understand the signal routings of a complex mixer setup.
f Tip: Clicking on any
of the input routings in the TV display highlights the corresponding mixer strip.
Output
Selecting the Output display view shows a graphic representation of the PatchMix DSP Mixer outputs. The output routings are divided into two categories: Physical Outputs and Host Outputs. Select either category by clicking on the Physical or Host button.
Physical Output Display Host Output Display
The Host Output display shows all the Insert Routings in addition to the Main Mix and Monitor out routings. Click on the desired row to make or break a physical output connection.
The Physical Output screen displays and allows you to connect the Main and Monitor outputs of the mixer to ā€œphysicalā€ analog or digital outputs. Click on the box in the mix or monitor area to make (or break) a connection.
The Host Output screen displays and allows you to view the Host (ASIO or WAVE) outputs of the mixer. See ā€œInsert Sectionā€ for information on how to connect the inserts.
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Auxiliary Effects & Returns

The section immediately below the TV Screen is where you assign the Auxiliary Effects. In a traditional mixing console, auxiliary effects sends are used to send part of the signal to outboard effect devices, then return the effected signal back into the mix using the effect returns. This is called a sidechain routing because the aux signal takes a detour through the effects before being summed back into the main mix.
Sidechain effects are usually effects that you might want applied to several channels, such as reverb. Effects such as EQ and compressors are usually NOT used as sidechain effects because they can cause unpredictable results when returned to the main bus.
4 - The PatchMix DSP Mixer
Main Section
f The Wet/Dry mix
setting in the effect should normally be set to 100% when the effect is inserted as a sidechain effect. This is because the Aux Return Amount will control the wet/dry mix.
Send
Amount
Sidechain
Effects
Return
Amount
Input
Pan
Fader
Mute
Aux Amt
Input
Aux Amt
Aux Bus
Sidechain Diagram
(Post-Fader Aux Sends)
Side
Return
Amount
Output
Send
Amount
Chain
Main Bus
You can also use the Auxiliary Sends as two extra mix buses. By turning the Aux Return amount all the way down and dropping an Insert Send into the chain, you can send the Auxiliary bus to any output you wish. See ā€œInsert Sectionā€ for more information.ā€š

Sync/Sample Rate Indicators

The Sync/Sample rate Indicators show the current session’s sample rate and whether it is internal or slaving to an external source. The display indicates which sample rate is currently in effect. If an external source is being used, the Source display reads ā€œEXTERNALā€.
When slaving to an external master source, the clock may drift slightly or change dramatically (i.e. abrupt sample rate change or unplugging of physical master source). PatchMix DSP is tolerant to minor drifting within the supported rates of 44.1k, 48k, 88.2k, 96k, 176.4k and 192k, but if the sample rate drifts out of this range the ā€œLOCKEDā€ LED will extinguish.
If the external clock source makes a radical sample rate change from the lower rates of
44.1k/48k to a higher rate or between any of the higher rates, the hardware automatiĀ­cally switches to internal 48kHz clock until the proper external clock is restored. The ā€œLOCKEDā€ LED will be off and the two units are NOT synchronized. Always check the ā€œLOCKEDā€ LED when using an external clock source to make sure you are sampleĀ­locked.
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4 - The PatchMix DSP Mixer Main Section

Output Section

Clip Indicators
Main
Insert
Section
Main Output Level Fader
Output Level
Meters
Sync/Sample
Rate Indicators
Monitor
Mute
Monitor Balance
Monitor Volume
Main Inserts
The main inserts allow you to apply effects to the main stereo signal coming out of the mixer (both mains and monitor). You might want to apply EQ or a compressor here. These inserts work just like the other insert locations—just drag and drop effects from the palette or right-click and add Sends, Sends/Returns. etc. Refer to the Mixer Block
Diagram
Main Output Fader
The main output fader controls the level of the main output (and the Monitor output as well since it is downstream from this control). The normal setting for this control is at unity or 0dB, but the control allows you to add up to +12dB of gain. High output levels may cause clipping on outboard amplifiers or other equipment.
MAIN MIX
10
10
20
20
30
30
0dB
-12dB
Output Level Meters
This stereo bar-graph meter reflects the digital level at the output of the mixer. The topmost red bar represents 0 dB or a full-scale digital signal. The peaks hold for a moment so that short transients can be monitored. Each bar = 1dB.
40
40
50
50
LR
Monitor Output Level
This control adjusts the monitor output level. Keep in mind that since the monitor level control comes after the Main Output Fader, nothing will be heard from your monitors if the main level is turned down.
Monitor Balance Control
This control sets the relative volume of the stereo monitor outputs and works just like the balance control on your home music system. This control is primarily used to make the volume from each speaker sound equal if you are not sitting exactly in the center of the two speakers.
f Tip: The volume
control on a multimedia computer keyboard can be used to control the Monitor Output Level on PatchMix.
Monitor Output Mute
This button completely cuts off the monitor output and provides a convenient way to instantly kill all sound without having to re-adjust the monitor level later. When the telephone rings, just hit the monitor mute to cut the noise.
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5 - Effects

Overview

PatchMix DSP comes complete with a host of great core DSP effects including Compressors, Delays, Choruses, Flangers and Reverb. Each 32-bit effect has various parameters for editing, as well as factory presets. You can also create and save as many of your own effect presets as you wish.
Since the effects are implemented in hardware, they don’t place any load on your host computer. This allows your valuable CPU cycles to be used for other applications or software plug-ins. The effects are only available at the 44.1 and 48kHz sample rates.
There is a finite limit to how many effects you can use at the same time. As you use up the PatchMix DSP resources, certain effects will appear ā€œgrayed outā€ and cannot be added to the mixer. Complex effects such as reverb use more DSP resources than say a 1-Band EQ. If you continue to add effects, all of the DSP resources will eventually be used up.

The Effects Palette

Click the FX button on the toolbar to bring up the Effects Palette. The Effects Palette contains two types of folders. The ā€œCore Effectsā€ folder contains the effect algorithms themselves. This folder cannot be modified. The other folders contain ā€œEffects Chainsā€, consisting of two or more effects grouped together. You can also add, delete, or modify Effects Chains and the folders that contain them. For more information on Effects Chains, see ā€œFX Insert Chainsā€ on page 54.
5 - Effects Overview
f Saving a session
ā€œdefragmentsā€ the effect/ DSP resources. If you have used all your effects and need another, try saving the session.
New Folder button Import/Export FX Button
Effect Categories
Core Effects
(Single Effect)
Multi-Effects
(Effect Combinations)
Distortion Lo-fi
Drums & Percussion
Environment
Equalization
Guitar
Multi Effects
Reverb
Synths & Keys
Vocal
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5 - Effects The Effects Palette
ī˜‚ To Select an Effect
1. Click the FX button to bring up the Effects Palette. The effect palette contains
numerous folders containing effects presets. Click on any folder to open it.
2. Select the effect you wish to use by clicking on it with the left mouse button and
while continuing to hold the mouse button, drag the effect into the desired location on the PatchMix DSP mixer screen and release the mouse button. Multi effects contain several effects along with their parameter settings.
3.
If you want to change the order of effects, simply Left-click and drag the effect to the desired location. Drag the effect to the area above or below the final destination and release the mouse button to move the effect.
ī˜‚ To Edit an Effect
1. Click on the Insert Location containing the effect you wish to edit. The effect
controls now appear on the TV screen.
2. Edit the effect parameters as desired.
ī˜‚ To Delete an Effect
1. Right-click on the Insert location containing the effect you wish to delete and a pop-
up list appears.
2. Select ā€œDelete Insert(s)ā€ from the top of the list. The effect will be deleted.
f The order of effects in
a chain can have a big effect on the sound.
This icon will appear when you drag an effect to a new location.

FX Insert Chains

FX Insert Chains can be used to save several effects and their settings into a single multiĀ­effect. When an effects chain is selected and dropped into an insert location, all the effects with control settings are copied as a single entity. Once dropped into an insert location, the effects are totally separate just as if you had placed them individually.
ī˜‚ To Save FX Insert Chains
1. Select two or more effects and place them into any consecutive insert locations.
2.
Set the effect parameters the way you want them, including wet/dry mix settings.
3. Right-click to bring up the list of options.
4. Select ā€œSave FX Insert Chainā€. The New FX preset dialog box appears.
5. Select a category folder where your preset will be placed, and enter a new preset
name for your FX Chain.
f Trim pots, peak meters
and test tone generators will also be included in the FX chain.
6. Select a folder where your new preset will be placed, then type in a new preset name
and click OK. Your preset is now saved.
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Creating, Renaming & Deleting Categories or Presets
There are several utilities to help you organize your effects presets.
ī˜‚ To Create a New Preset Category
You can create your own category folders to help organize your effects presets.
1. Left-click on the New Folder icon at the top of the Effects Palette. A pop-up dialog
box appears asking you to ā€œEnter the Name of the New Category.ā€
• Alternatively, you can Right-click over an Effects Folder, which calls a pop-up dialog box with the option to ā€œCreate New Category.ā€
2. Type in a name for your new folder.
3. Click OK to create a new folder or Cancel to cancel the operation.
ī˜‚ To Delete an Effect Category or Preset
1. Right-click on the category folder you wish to delete. A pop-up selection box
appears.
2. Select ā€œDelete Categoryā€. A popup dialog box appears warning you that this action
will delete all presets in the folder.
3. Click OK to delete the folder or Cancel to cancel the operation.
5 - Effects
The Effects Palette
ī˜‚ To Rename an Effects Category
1. Right-click on the category folder you wish to rename. A pop-up selection box
appears.
2. Select ā€œRename Categoryā€. A pop-up dialog box appears, asking you to ā€œEnter New
Category Name.ā€
3. Click OK to rename the folder or Cancel to cancel the operation.
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5 - Effects The Effects Palette
Importing and Exporting Core FX Presets and FX Insert Chains
These utilities make it easy to import or export your FX Presets and FX Insert Chains. You can share presets with your friends or download new presets from the Internet.
ī˜‚ To Import Core FX Presets
This option imports complete folders of Core FX presets into the E-MU PatchMix DSP folder (normally located here: ā€œC:\Program Files\Creative Professional\E-MU PatchMix DSP\Core Effectsā€). If the name of an imported FX preset exactly matches a preset you already have, a number will be appended to end of the imported preset name.
1. Click the Import/Export FX Library button from the FX Palette.
2. Select Import FX Library. The ā€œBrowse for Folderā€ window appears.
3. Choose the folder where the Core FX presets you wish to import are located.
4. The selected folder of Core FX presets will be copied into the Core Effects folder of
PatchMix DSP.
ī˜‚ To Import FX Category Folders
This option imports complete category folders of FX Chains into the E-MU PatchMix DSP folder (normally located here: ā€œC:\Program Files\Creative Professional\E-MU PatchMix DSP\Effect Presetsā€). If the name of an imported FX preset exactly matches a preset you already have, a number will be appended to end of the imported preset name.
1. Click the Import/Export FX Library button from the FX Palette.
2. Select Import FX Category. The ā€œBrowse for Folderā€ window appears.
3. Choose the folder where the FX Chains you wish to import are located.
4. The selected folder of FX Chains will be copied into the Effect Presets folder of
PatchMix DSP.
ī˜‚ To Export your Core FX Presets
This option exports your Core FX presets to a folder of your choice.
1. Click the Import/Export FX Library button from the FX Palette.
2. Select Export FX Library. The ā€œBrowse for Folderā€ window appears.
3. Choose a destination location for the Core FX presets, then press OK.
4. The Core FX presets will be copied to the selected destination.
ī˜‚ To Export your FX Category Folders
This option exports a single category of FX chains to a folder of your choice.
1. Click the Import/Export FX Library button from the FX Palette.
2. Select Export FX Category. A pop-up dialog box appears asking you to ā€œChoose the
FX Category to be exportedā€.
3. Choose the desired FX Category to export. Press OK to continue or Cancel to
cancel the operation.
4. The ā€œBrowse for Folderā€ window appears. Choose a destination location for the
Core FX presets, then press OK.
5. The FX Chains will be copied to the selected destination.
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FX Edit Screen

Click on an FX Insert to display the parameters for that effect. If an insert effect is not selected, the FX display will read ā€œNo Insertā€.
Most effects have a wet/dry mix parameter to control the ratio of effect-to-plain signal. The wet/dry setting is stored with the FX preset. The effect parameters vary with the type of effect. Generally if an effect is placed in an Aux Send, the wet/dry mix in the effect should be set to 100% wet since the Aux Return amount controls how much effect is applied.
The User Preset section is located at the bottom of the FX Edit screen. User presets are variations of the main effect and can be edited, deleted, renamed or overwritten as you wish.
5 - Effects
FX Edit Screen
E Note: Effects have to
be placed into an insert location before you can program them.
Effects Display
View Button
Wet/Dry Mix Control
ī˜‚ To Bypass an Insert:
Effect Location
Effect Bypass & Solo Buttons
Effect Parameters
User Preset Section
Inserts can be bypassed if you want to temporarily hear the audio without the effect or insert. Bypass can also be used to turn off a Send Insert.
Method #1
1. Click on the Effect (in the Insert section)
2. Click the Bypass button in the TV display.
Method #2
1. Right-click over the Insert you want to bypass (in the Insert section). A pop-up
menu appears.
2. Select ā€œBypass Insertā€ from the list of options. The insert effect name will ā€œgray-outā€
to indicate that the insert effect is bypassed.
ī˜‚ To Solo an Insert:
Inserts can also be soloed. Solo bypasses all the other inserts in the strip and allows you to hear only the soloed effect. This feature is very useful when adjusting the effect parameters.
Method #1
1. Click on the Insert Effect (in the Insert section).
2. Click the Solo button in the TV display.
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5 - Effects FX Edit Screen
Method #2
1. Right-click over the Insert Effect you want to Solo (in the Insert section). A pop-up
menu appears.
2. Select ā€œSolo Insertā€ from the list of options. The other Insert Effect names in the
strip will ā€œgray-outā€ to indicate that they are bypassed.
ī˜‚ To Bypass ALL
All the inserts in a strip can be bypassed with a single command.
1. Right-click over any Effect in the Insert section. A pop-up menu appears.
2. Select ā€œBypass All Insertsā€ from the list of options. All the insert names will be
ā€œgrayed-outā€ to indicate that they are bypassed.
ī˜‚ To Un-Bypass ALL
All the inserts in a strip can also be un-bypassed with a single command. This command works even if only some of the effects are bypassed.
1. Right-click over any Effect in the Insert section. A pop-up menu appears.
2. Select ā€œUn-Bypass All Insertsā€ from the list of options. All the insert names will light
to indicate that they are active.

User Preset Section

Each core effect has a set of User Presets, that you can use to store your favorite effect parameter settings. We’ve included a good collection of user presets to get you started. The user presets are accessed from the bar at the bottom of the TV screen. The user preset edit menu allows you to select stored presets, create new presets, rename or delete existing presets, or overwrite existing presets with your modified settings. User presets stay with the Mixer application regardless of which Session is open.
Click here for Edit Menu
Click here to Select Presets
ī˜‚ To Select a User Preset
1. Select the FX display in the TV screen.
2. Select the desired insert effect, highlighting it. The effect parameters appear in the TV
screen.
3. Click on the icon on the preset menu. A drop-down preset list appears.
4. Select a preset from the list.
ī˜‚ To Create a New User Preset
1. Select the FX display in the TV screen.
2. Select the desired insert effect, highlighting it. The effect parameters appear in the TV
screen.
3. Click on the Edit button. A pop-up menu appears.
4. Select New. A pop-up dialog box appears asking you to name the new preset.
5. Name the preset and click OK. Your new preset is now saved.
E To copy or share User
Presets, you must save them as FX Palette effects.
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ī˜‚ To Delete a User Preset
1. Select the user preset you wish to delete from the user preset menu.
2. Click on the Edit button. A pop-up menu appears.
3. Select Delete. A pop-up dialog box appears asking you to confirm your action.
4. Click OK to delete the preset or No or Cancel to cancel the operation.
ī˜‚ To Rename a User Preset
1. Select the user preset you wish to rename from the user preset menu.
2. Click on the Edit button. A pop-up menu appears.
3. Select Rename. A pop-up dialog box appears asking you to rename the preset.
4. Type in the new preset name, then click OK to rename the preset or Cancel to cancel
the operation.
ī˜‚ To Overwrite or Save a User Preset
This operation allows you to overwrite an existing preset with a newer version.
1. Select the user preset you wish to modify from the user preset menu and make any
changes you wish.
2. Click on the Edit button. A pop-up menu appears.
3. Select Overwrite/Save. The current preset will be overwritten with the new settings.
5 - Effects
FX Edit Screen

Core Effects and Effects Presets

The Core Effects cannot be removed or copied. Effect presets (stored in ā€œC:\Program Files\Creative Professional\E-MU 1616\E-MU PatchMix DSP\Effect Presetsā€) can be copied, e-mailed or shared like any other computer file.
f Hint: You can open
the effects presets with ā€œNotePadā€ or other word processor to view and edit the name and parameters.
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5 - Effects List of Core Effects

List of Core Effects

Stereo Reverb Frequency Shifter Mono Delay 750
Lite Reverb Auto-Wah Mono Delay 1500
Compressor Vocal Morpher Mono Delay 3000
Leveling Amp 1-Band Para EQ Stereo Delay 100
Chorus 1-Band Shelf EQ Stereo Delay 250
Flanger 3-Band EQ Stereo Delay 500
Distortion 4-Band EQ Stereo Delay 750
Speaker Sim Mono Delay 100 Stereo Delay 1500
Rotary Mono Delay 250
Phase Shifter Mono Delay 500

DSP Resource Usage

There are two main factors which determine the total number of effects available for use at any given time: Tank Memory and DSP Instructions. Using too much of either resource will cause effects to be unavailable (grayed out) in the FX menu. In addition, the strips themselves use DSP Instructions, so only create strips that you actually need.
Tank memory is the memory used by delay-based effects such as reverb and digital delays. All the reverbs and delays aside from the Mono Delay 100 and Stereo Delay 100 use varying amounts of tank memory.
The DSP instructions are used by all the effects. Effects with multiple stages, such as multi-band EQs or the speaker simulator use more DSP instructions than a 1-Band EQ.
Tank memory tends to get used first, and so we’ve provided many delay line effects to allow maximum conservation of this precious resource. Use only the longest delay you actually need.
f Tip: Saving a session
ā€œdefragmentsā€ the effect/ DSP resources. If you have used all your effects and need another, try saving the session.
The chart below shows three possible effects combinations. These were created by using up the reverb resources first. Even more simultaneous effects are possible if fewer reverbs and shorter delays are used.
Examples of Effects Usage
(with a WAVE, ASIO Return & 2 Inputs)
Example 1 No. Example 2 No. Example 3 No.
Stereo Reverb 2 Lite Reverb 5 Stereo Reverb 1
4-Band EQ 4 3-Band EQ 5 Lite Reverb 2
3-Band EQ 2 1-Band EQ 4 Stereo Delay 1500 1
1-Band EQ 6 Compressor 1 Mono Delay 250 1
Compressor 6 Mono Delay 1500 1 Compressor 6
Chorus 1 Mono Delay 250 1 Chorus 2
Mono Delay 1500 1 Auto-Wah 1 Flanger 2
4-Band EQ 3
3-Band EQ 3
Total Effects 22 Total Effects 18 Total Effects 21
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Core Effects Descriptions

f

1-Band Para EQ

+15dB
Boost
Width
+
Gain
-
Cut
-15dB
Center
Frequency
Parameter Description
Gain Sets the amount of cut (-) or boost (+) of the selected frequency
band. Range: -15dB to +15dB
Center Frequency Sets the range of frequencies to be cut or boosted with the Gain
control. Range: 80Hz to 16kHz
Bandwidth Sets the width of the frequency range for the Center Frequency
band that will be cut or boosted by the Gain control. Range: 1semitone to 36 semitones
This single band parametric equalizer is useful when you just want to boost or cut a single range of frequencies. For example, if you just want to brighten up the lead vocal a bit, you might choose this EQ. This EQ offers up to ±15dB cut or boost.
5 - Effects
Core Effects Descriptions

1-Band Shelf EQ

This single band shelving equalizer is useful when you just want to boost or cut a single range of frequencies at the high or low end of the spectrum. For example, if you just want to add a little more bass, there’s no need to waste a 3-band EQ. Just choose low shelf, then adjust the gain and frequency. This EQ offers up to ±15dB cut or boost.
Low Shelf or… High Shel
+15dB
Corner
Freq
Corner
Freq
+
Boost
Gain
-
Cut
-15dB
Parameter Description
Shelf Type Allows you to choose either low shelving or high shelving EQ.
Gain Sets the amount of cut (-) or boost (+) of the shelf.
Corner Frequency Sets the frequency where the signal begins getting cut or boosted
Frequency
Range: -15dB to +15dB
with the Gain control. Range: 80Hz to 16kHz
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5 - Effects
f
Core Effects Descriptions

3-Band EQ

This versatile equalizer provides two shelving filters at the high and low ends of the frequency range and a fully parametric band in the center. Up to ±24 dB of boost or cut is provided for each band.
Low Shelf Mid Band High Shel
+24dB
Corner
Freq.
Corner
Freq.
E Note: The Wet/Dry
Mix control on an equalizer should normally be set to 100% wet or unpredictable results may occur.
Width
Center
Boost
Gain
Cut
+
-
-24dB
Frequency
ī˜‚ Setting up a Parametric EQ
1. Turn up the gain on the band you are working with. This allows you to easily hear
the effect of the filter.
2. Reduce the bandwidth if you are working with a mid-band.
3. Adjust the Center Frequency to ā€œzero-inā€ on the frequencies you wish to boost/cut.
4. Set the Gain to a positive value to boost frequencies or to a negative value to cut out
frequencies.
5. Widen the Bandwidth to create a more natural sound.
6. Adjust and tweak as needed.
Parameter Description
High Shelf Gain Sets the amount of cut (-) or boost (+) of the high frequency shelf.
Range: -24dB to +24dB
High Corner Freq. Sets the frequency where the signal begins getting cut or boosted
with the High Gain control. Range: 4kHz to 16kHz
Mid Gain Sets the amount of cut (-) or boost (+) of the mid frequency band.
Range: -24dB to +24dB
Mid Freq. 1 Sets the range of frequencies to be cut or boosted with the Mid
Gain control. Range: 200Hz to 3kHz
Mid Bandwidth Sets the width of the frequency range for the Mid Center
Frequency band that will be cut or boosted by the Mid Gain control. Range: 1 semitone to 1 octave
Low Shelf Gain Sets the amount of cut (-) or boost (+) of the low frequency shelf.
Range: -24dB to +24dB
Low Corner Freq. Sets the frequency where the signal begins getting cut or boosted
with the Low Gain control. Range: 50Hz to 800Hz
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4-Band EQ

This 4-band equalizer provides two shelving filters at the high and low ends of the frequency range and two fully parametric bands in the center. Up to ±24 dB of boost or cut is provided for each band.
Note: The Wet/Dry Mix control on an equalizer should normally be set to 100% wet or unpredictable results may occur.
For more information about setting up a parametric EQ, see page 62.
5 - Effects
Core Effects Descriptions
Low-Shelf Mid 1-Band High-Shelf
Corner
Frequency
Width
Center
Frequency
Boost
Gain
Cut
+
-
Mid 2-Band
Corner
Frequency
Width
Center
Frequency
Frequency
Parameter Description
High Shelf Gain Sets the amount of cut (-) or boost (+) of the high frequency shelf.
Range: -24dB to +24dB
High Corner Freq. Sets the frequency where the signal begins getting cut or
boosted with the High Gain control. Range: 4kHz to 16kHz
Mid 2 Gain Sets the amount of cut (-) or boost (+) of the Mid 2 Frequency
band. Range: -24dB to +24dB
Mid 2 Center Freq. Sets the range of frequencies to be cut or boosted with the Mid 2
Gain control. Range: 1kHz to 8kHz
Mid 2 Bandwidth Sets the width of the frequency range for the Mid 2 Center
Frequency band that will be cut or boosted by the Mid 2 Gain control. Range: .01 octave to 1 octave
Mid 1 Gain Sets the amount of cut (-) or boost (+) of the Mid 1 Frequency
band. Range: -24dB to +24dB
Mid 1 Center Freq. Sets the range of frequencies to be cut or boosted with the Mid 1
Gain control. Range: 200Hz to 3kHz
Mid 1 Bandwidth Sets the width of the frequency range for the Mid 1 Center
Frequency band that will be cut or boosted by the Mid 1 Gain control. Range: .01 octave to 1 octave
Low Shelf Gain Sets the amount of cut (-) or boost (+) of the low frequency shelf.
Range: -24dB to +24dB
Low Corner Freq. Sets the frequency where the signal begins getting cut or
boosted with the Low Gain control. Range: 50Hz to 800Hz
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5 - Effects Core Effects Descriptions

Auto-Wah

This effect creates the sound of a guitar wah-wah pedal. The ā€œWahā€ filter sweep is automatically triggered from the amplitude envelope of the input sound. Auto-wah works well with percussive sounds such as guitar or bass.
The Auto-Wah is a bandpass filter whose frequency can be swept up or down by an envelope follower, which extracts the volume contour of the input signal. The Envelope Sensitivity setting allows you to properly set up the envelope follower to receive a wide variety of input signals. This ā€œenvelopeā€, or volume contour, controls the frequency of the bandpass filter so that it sweeps up and down with each new note. The Attack controls the rate of the note-on sweep. As the input sound fades away, the filter sweeps back at a rate determined by the Release setting.
The wah direction allows the filter to be swept either up or down in frequency. Use a higher Center Frequency setting when the wah direction is down.
Auto-Wah Filter
Center
Frequency
Bandwidth
Envelope Sensitivity
Input Wave
Sweep Range
Attack Release
Envelope Follower
Parameter Description
Wah Direction Allows you to sweep the wah up or down.
Env. Sensitivity Controls how closely the wah sweep follows the input signal.
Range: -12dB to +18dB
Env. Attack Time Sets the starting rate of the ā€œwahā€ sweep.
Range: 0ms to 500ms
Env. Release Time Sets the ending or release rate of the ā€œwahā€ sweep.
Range: 10ms to 1000ms
Sweep Range Controls the amount of ā€œwahā€ sweep. Range: 0% to 100%
Center Frequency Sets the initial bandpass filter frequency.
Range: 80Hz to 2400Hz
Bandwidth Sets the width of the bandpass filter. Range: 1Hz to 800Hz
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Chorus

t
An audio delay in the range of 15-20 milliseconds is too short to be an echo, but is perceived by the ear as a distinctly separate sound. If we now vary the delay time in this range, an effect called chorus is created, which gives the illusion of multiple sound sources. A slight amount of feedback serves to increase the effect. A very slow LFO rate is usually best for a realistic effect, but a faster LFO rate can also be useful with minimal LFO depth (.2). Since this is a stereo chorus, an LFO phase parameter is included which can be used to widen the stereo image.
Parameter Description
Delay Sets the length of the delay. Range: 0ms to 20ms.
Feedback Sets the amount of delayed signal that will be recirculated through
the delay line. Range: 0% to 100%
LFO Rate Sets the frequency of the low frequency oscillator.
Range: .01Hz to 10Hz
LFO Depth Sets how much the LFO affects the delay time. Increases the
animation and amount of the chorus effect. Range: 0% to 100%
LFO Waveform Selectable between Sine or Triangle wave.
LFO L/R Phase Controls the stereo width by adjusting the phase difference of the LFO
waveform between left and right channels. Range: -180° to +180°
5 - Effects
Core Effects Descriptions

Compressor

In its simplest form, an audio compressor is just an automatic gain control. When the volume gets too loud, the compressor automatically turns it down. Compressors are useful in musical applications because they allow you to record a ā€œhotterā€ signal without overloading the recording device.
Since the compressor turns down the gain of the signal, you might wonder how can it make the signal level stronger. A Post Gain control allows you to boost the output gain of the compressor in order to make up for the gain reduction. The overall level is higher and only turned down when the signal level gets too loud. This level is called the Threshold, which just happens to be the most important control on the compressor.
Signal path = Stereo
In
Delay
Level
VCA
Ou
Control
Threshold
Ratio
E-MU PCI Digital Audio System 65
Attack
Release
Post Gain
Page 66
5 - Effects Core Effects Descriptions
Basic Controls
The three main controls of a compressor are the Ratio control, the Threshold control and the Gain control.
If the signal falls below the Threshold, no processing will take place. Signals exceeding the Threshold will have gain reduction applied as set by the ratio control. This important control allows you to dial in the range of amplitudes you want to tame. For example, if you’re trying to trim off just the loudest peaks, set the threshold so the gain reduction meter only shows compression during these peaks. One of the biggest mistakes in using a compressor is having the threshold set too low. This adds noise as the compressor will always be reducing the volume.
The Ratio control determines how strongly the compressor will affect the signal. The higher the ratio, the more reduction will be applied. If the ratio is high enough, (above 10:1) the signal will effectively be prevented from getting any louder. In this situation, the compressor will be acting as a Limiter, placing an upper limit on the signal level. In general, ratios from 2:1 to 6:1 are considered compression and higher ratios above 10:1 are considered limiting.
The Post Gain control amplifies the signal after it has been compressed to bring it back up in volume. If you don’t increase the gain, the compressed signal will be much lower in volume.
Two other important controls are Attack and Release. Attack controls how quickly the gain is turned down after the signal exceeds the threshold. Release controls how fast the gain is returned to its normal setting after the signal has fallen below the threshold again. An attack setting of about 10 milliseconds will delay the onset of compression long enough to preserve the attack transients in guitar, bass or drums while allowing the sustain portion of the sound to be compressed. Longer release times are generally used to reduce the so called ā€œpumpingā€ effect as the compressor turns on and off. Don’t make the release time too long, however, or the compressor won’t have time to recover for the next pluck or hit. In general, the attack and release controls are used to smooth out the action of the compressor, but they can also be used to create special effects.
The Pre-Delay parameter lets the level detector ā€œlook into the futureā€ up to 4 milli-
seconds in order to anticipate upcoming peaks in the signal. This is accomplished of course, by inserting delay into the signal path. This lookahead technique allows the use of slower attack times without missing signal peaks. This parameter is especially effective on drums and percussion.
The Input Meter allows you to monitor the strength of your input signal. Always try to boost the signal before the compressor if you can.
The Compression Meter shows the amount of gain reduction being applied. Since this meter displays how much the gain is being turned down, the meter moves from right to left, instead of left to right like a normal meter.
Parameter Description
Threshold Threshold sets the input signal level above which dynamic range
compression takes place. Everything above the threshold will be brought down in volume. Range: -60dB to +12dB
Ratio Sets the ratio of input signal level to output signal level, or
ā€œhow muchā€ compression will be applied. Range: 1:1 to āˆž:1
Post Gain Amplifies the signal after it has been compressed to bring up the
volume. Range -60dB to +60dB
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Parameter Description
t
Attack Time Controls how quickly the gain is turned down after the signal
exceeds the threshold. Range .1ms to 500ms
Release Time Controls how fast the gain is returned to its normal setting after
the signal has fallen below the threshold. Range: 50ms to 3000ms
Pre-Delay Allows the use of slower attack times without missing signal peaks.
Range: 0ms to 3 ms
Input Meter Allows you to monitor the strength of the input signal.
Gain Reduction Meter Shows the amount of gain reduction being applied.

Distortion

Most audio processors aim to provide low distortion, but not this one! The sole purpose of this effect is to add distortion, and lots of it. This effect provides ā€œfuzz boxā€ style, clipping distortion which is particularly effective on guitar, bass, organs, electric pianos or whatever.
The input signal first passes through a lowpass filter. The Lowpass Filter Cutoff Frequency allows you to control the number of new harmonics that will be generated by the distortion element. The distortion element has an Edge control which controls ā€œhow muchā€ distortion will be added. A bandpass filter follows the distortion generator. The EQ Center control lets you select a particular band of frequencies to be output. The EQ Bandwidth controls the width of the center frequency band. Finally, a gain control allows you to make up for any gain loss through the effect.
Use the Wet/Dry mix control in conjunction with the Edge control to reduce the amount of distortion, or go wild and turn everything to 11!
5 - Effects
Core Effects Descriptions
Lowpass
Filter
In
Signal path = Stereo
Distortion
LP Filter
Cutoff
Parameter Description
Pre EQ LP Cutoff Controls the amount of high frequency audio admitted to
the distortion. Range: 80Hz to 24kHz
Edge Sets the amount of distortion and new harmonics
generated. Range: 0-100
Gain Sets the output volume of the effect. Range: -60dB to 0dB
Post EQ Center Freq. Sets the frequency of the output bandpass filter.
Range: 80Hz to 24kHz
Post EQ Bandwidth Sets the width of the output bandpass filter.
Range: 80Hz to 24kHz
Edge
Bandpass
Filter
Ou
GainEQ BW
EQ Center
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5 - Effects
t
Core Effects Descriptions

Flanger

A flanger is a very short delay line whose output is mixed back together with the original sound. Mixing the original and delayed signals results in multiple frequency cancellaĀ­tions known as a comb filter. Since the flanger is a type of filter, it works best with harmonically rich sounds.
A low frequency oscillator is included to slowly change the delay time. This creates a rich, sweeping effect as the notches move up and down across the frequency range. The amount of feedback deepens the notches, intensifying the effect. You can invert the feedback signal by choosing a negative feedback value. Inverting the feedback signal creates peaks in the notch filter and deepens the effect.
Feedback
In
Flanger
Signal path = Stereo
Delay
Parameter Description
Delay Sets the initial delay of the flanger in .01 millisecond increments.
This parameter allows you to ā€œtuneā€ the flanger to a specific frequency range. Range: .01ms to 4ms
Feedback Controls how much signal is recirculated through the delay line
and increases resonance. Negative values can produce intense flanging with some signals. Range 0% to 100%
LFO Rate Sets the speed of the flanger sweep. Range: .01 Hz to 10Hz
LFO Depth Sets how much the LFO affects the delay time. Increases the
animation and amount of the flanging effect. Range 05 to 100%
LFO Waveform Selectable between Sine or Triangle wave.
LFO L/R Phase Controls the stereo width by adjusting the phase difference
between the left and right sweeps. Range: -180° to +180°
LFO
Waveform
Phase
Ou
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Freq Shifter

This unusual effect is sometimes called ā€œspectrum shiftingā€ or ā€œsingle sideband modulationā€. Frequency shifting shifts every frequency in the signal by a fixed number of Hz which causes the harmonics to lose their normal relationship. The more common pitch shifter, in contrast, preserves the harmonic relationships of the signal and so is better suited to creating ā€œmusicalā€ harmonies.
This isn’t to say that the frequency shifter can’t be used musically. Small intervals of frequency shifting (1 Hz and below) can produce a wonderful, lush chorusing or phasing effect. For bizarre frequency shifting effects, simply crank up the frequency knob. Frequencies can be shifted up or down by any specified amount from .1 Hz to 24 kHz. You can also shift pitch up on one side and down on the other if you wish.
Comparison between Pitch and Frequency Shifting
5 - Effects
Core Effects Descriptions
f You can also type in
exact frequencies to a resolution of 1/10 Hz.
Harmonic
Original
(Hz)
Pitch Shifted
(100 Hz)
Frequency Shifted
(100 Hz)
1 200 300 300
2 400 600 500
3 600 900 700
4 800 1200 900
5 1000 1500 1100
6 1200 1800 1300
7 1400 2100 1500
8 1600 2400 1700
Parameter Description
Frequency Sets the number of Hz that will be added or subtracted with every
harmonic in the signal. Range: .01Hz to 24kHz
Left Direction Sets pitch shift up or down for the left channel.
Right Direction Sets pitch shift up or down for the right channel.
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5 - Effects Core Effects Descriptions

Leveling Amp

The first compressors developed in the 1950’s were based on a slow-acting optical gain cells which were able to control the signal level in a very subtle and musical way. This effect is a digital recreation of the leveling amps of yesteryear.
The leveling amp uses a large amount of ā€œlookahead delayā€ to apply gentle gain reduction. Because of this delay, the leveling amp is not suitable for applications which require realtime monitoring of the signal. This smooth and gentle compressor is designed to be used in situations where delay does not pose a problem, such as mastering a mix or compressing prerecorded stereo material.
Post Gain is the only control on the leveling amp. This control is used to make up the volume lost by the compression. The Compression Ratio is fixed at about 2.5:1. If a large peak is detected, the effect will automatically increase the compression ratio to keep the audio output controlled.
The gain reduction meter shows you how much gain reduction is being applied. Since the gain reduction meter displays how much the gain is being turned down, the meter moves from right to left, instead of left to right like most meters.
Post Gain Amplifies the signal after it has been compressed to
bring up the volume. Range 0dB to 36dB
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Lite Reverb

Reverberation is a simulation of a natural space such as a room or hall. The Lite Reverb algorithm is designed to simulate various rooms and reverberation plates while using fewer DSP resources than the Stereo Reverb. Up to five Lite Reverbs can be used at once.
Decay time defines the time it takes for the reflected sound from the room to decay or die away. The diagram below shows a generalized reverberation envelope.
5 - Effects
Core Effects Descriptions
Early Reflections
After a short pre-delay period, the echoes from the closest walls or ceiling are heard. These first echoes, or Early Reflections, vary greatly depending on the type of room. Some time after the early reflection cluster ends, the actual Reverberation (a dense cloud of complex wall reflections) begins and decays according to the time set by the Decay Time parameter. The Reverberance parameter controls the density and smearing of both the early reflections and the reverberation cloud.
High frequency energy tends to fade away first as a sound is dissipated in a room. The High Frequency Decay Factor adjusts the time it takes for the high frequency energy to die away and thus changes the characteristics of the room. Rooms with smooth, hard surfaces are more reflective and have less high frequency damping. Rooms filled with sound absorbing materials, such as curtains or people, have more high frequency damping.
The Low Frequency Decay Factor parameter adjusts the time it takes for the low frequencies to die away. This control adjusts the ā€œboominessā€ of the room.
Parameter Description
Decay Time Sets the reverb decay time. Range: 0% to 100%
HF Decay Factor Sets the rate at which high frequencies die away. The high
LF Decay Factor Sets the rate at which low frequencies die away. The low
Early Reflections Sets the volume of the initial wall reflections.
Reverberance Sets the amount of scattering of the early reflections and
Reverberation
frequencies last longer as the percentage is increased. Range: 0% to 100%
frequencies last longer as the percentage is increased. Range: 0% to 100%
Range: 0% to 100%
the reverberation cloud. Range: 0% to 100%
Time
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5 - Effects
t
Core Effects Descriptions

Mono Delays - 100, 250, 500, 750, 1500, 3000

A delay line makes a copy of the incoming audio, holds it in memory, then plays it back after a predetermined time. The delay number refers to the maximum delay time that can be produced by the delay line. The six lengths, from 100 ms to 3 seconds, allow you to make the most efficient use of the effect memory resource.
Long delays produce echoes, short delays can be used for doubling or slapback effects. Very short delays can be used to produce resonant flanging and comb filter effects or create monotone robotic-sounding effects (Hint: use feedback). Stereo signals are summed together before entering the Mono Delay.
There is also a feedback path to send the delayed audio back through the delay line. When creating echo effects, the feedback controls how many echoes will be produced. With short delays, the feedback control acts as a resonance control, increasing the amount of comb filtering produced by the delay line. Comb fi
A High Frequency Rolloff filter in the feedback path cuts some of the high frequency energy each time the audio goes through the delay line. This simulates the natural absorption of high frequencies in a room and can also be used to simulate tape-based echo units.
The Wet/Dry mix controls how loud the echoes are in relation to the original signal.
ltering: See page 68.
Feedback
HF
Rolloff
L In
L Out
Delay
R In
Delay Time
Parameter Description
Delay Time
Mono Delay 100 Mono Delay 250 Mono Delay 500
Mono Delay 750 Mono Delay 1500 Mono Delay 3000
Feedback Sets the amount of delayed signal that will be recirculated through
High Freq. Rolloff Damps high frequencies in the feedback path.
Sets the length of the delay in milliseconds.
(.01ms. minimum increment between settings)
Range: 1 millisecond to 100 milliseconds Range: 1 millisecond to 250 milliseconds Range: 1 millisecond to 500 milliseconds Range: 1 millisecond to 750 milliseconds Range: 1 millisecond to 1.5 seconds Range: 1 millisecond to 3 seconds
the delay line. Range: 0% to 100%
Range: 0% to 100%
R Ou
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Phase Shifter

t
A phase shifter produces a fixed number of peaks and notches in the audio spectrum which can be swept up and down in frequency with a low frequency oscillator (LFO). This creates a swirly, ethereal sound with harmonically rich sound sources of a type of pitch shift with simpler sounds. The phase shifter was invented in the 1970’s and the characteristic sound of this device evokes emotions of that musical era.
By setting the LFO Depth to zero and tuning the LFO Center, a fixed multi-notch filter is created.
Feedback
In
Phase
Signal path = Stereo
Shifter
Ou
5 - Effects
Core Effects Descriptions
LFO Center
LFO
LFO Rate
Parameter Description
LFO Center Sets the initial offset of the LFO and changes the position of the
peaks and notches. Range: 0% to 100%
Feedback Increases the depth of the notches and height of the peaks.
Range: 0% to 100%
LFO Rate Controls the sweep rate of the Low Frequency Oscillator.
Range: .01Hz to 10Hz
LFO Depth Controls how much the Center Frequency is swept by the LFO.
Range: 0% to 100%
Waveform Selects a Sine or Triangle wave for the LFO
LFO L/R Phase Controls the stereo width by adjusting the phase difference
between the left and right sweeps. Range: -180° to +180°

Rotary

This is a simulation of a rotating speaker used on organs. The rotating speaker was invented to give static organ tones a pipe organ type of animation, but this distinctive sound became a legend in its own right. Spinning a sound around the room creates a doppler pitch shift along with many other complex and musically pleasing sonic effects.
The Rotary incorporates acceleration and deceleration as you switch between the two speeds.
Parameter Description
Speed Switches between slow or fast rotor speeds with
acceleration and deceleration as the speed changes.
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5 - Effects Core Effects Descriptions

Speaker Simulator

The Speaker Simulator provides realistic guitar speaker responses and is designed for use with guitar, bass or synthesizer. Twelve popular guitar amp speaker cabinets are modeled.
There is only one parameter on this effect. Just select the speaker you want and listen. Normally this effect should be used with the Mix control set to 100%.
Speaker Type Description
British Stack 1 & 2 Modeled from a British 8-speaker high power amplifier stack.
British Combo 1-3 Modeled from a British 2-speaker combo amplifier.
Tweed Combo 1-3 Modeled from an American, 1950’s era, 2-speaker combo amplifier.
2 x 12 Combo Modeled from an American, 1960’s era, 2-speaker combo amplifier.
4 x 12 Combo Modeled from an American, 1960’s era, 4-speaker amplifier set.
Metal Stack 1 & 2 Modeled from a modern era, power amplifier stack.
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Stereo Delays - 100, 250, 500, 750, 1500

t
The Stereo Delays are true stereo delay lines in that the left and right channels are kept entirely separate from each other. The delay number refers to the maximum delay time that can be produced by the delay lines. The five different lengths, from 100 ms to 1.5 seconds, allow you to make the most efficient use of the effect memory resource.
Because the left and right channels can have different delay times, you can create a panning effect by setting one delay long and the other short. Very short delay times combined with a high feedback amount can be used to create monotone roboticĀ­sounding effects. Using the longer stereo delays, you can ā€œoverdubā€ musical lines one on top of the other with the feedback control turned up.
Feedback
HF
Rolloff
In
5 - Effects
Core Effects Descriptions
Delay
Signal path = Stereo
L Delay
Time
Parameter Description
Left Delay Time Sets the length of the delay for the left channel in milliseconds.
Right Delay Time Sets the length of the delay for the right channel in milliseconds.
Delay Time (L & R) Stereo Delay 100 Stereo Delay 250 Stereo Delay 500
Stereo Delay 750 Stereo Delay 1500
Feedback Sets the amount of delayed signal that will be recirculated through
High Freq. Rolloff Damps high frequencies in the feedback path. Range: 0% to 100%
(.01ms. minimum increment between settings)
Range: 1 millisecond to 100 milliseconds Range: 1 millisecond to 250 milliseconds Range: 1 millisecond to 500 milliseconds Range: 1 millisecond to 750 milliseconds Range: 1 millisecond to 1.5 seconds
the delay line. Range: 0% to 100%
R Delay
Time
Ou
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5 - Effects Core Effects Descriptions
Stereo Reverb
Reverberation is a simulation of a natural space such as a room or hall. The stereo reverb algorithm is designed to simulate various halls, rooms and reverberation plates.
Decay time defines the time it takes for the reflected sound from the room to decay or die away. The diagram below shows a generalized reverberation envelope.
Time
Early Reflections
Late Reverb
After a short pre-delay period, the echoes from the closest walls or ceiling are heard. These first echoes, or early reflections, vary greatly depending on the type of room. Some time after the early reflection cluster ends (late reverb delay), the late reverberation (a dense cloud of complex wall reflections) begins and decays according to the time set by the Decay Time parameter.
Diffusion is the amount of scattering and density of the late reverberation cloud. Rooms with many complex surfaces have more diffusion than bare rooms.
High frequency energy tends to fade away first as a sound is dissipated in a room. The High Frequency Damping parameter adjusts the time it takes for the high frequency energy to die away and thus changes the characteristics of the room. Rooms with smooth, hard surfaces are more reflective and have less high frequency damping. Rooms filled with sound absorbing materials, such as curtains or people, have more high frequency damping.
The Low Frequency Damping parameter adjusts the time it takes for the low frequencies to die away. This control adjusts the ā€œboominessā€ of the room.
Parameter Description
Decay Time Sets the length of the Late Reverb. Range 1.5 to 30 seconds
Early Reflections Level Sets the volume of the initial wall reflections.
Range: 0% to 100%
Early/Late Reverb Bal Adjusts the balance between early refections and late reverb.
Range: 0% to 100%
Late Reverb Delay Sets the time between early reflections and the onset of the late
reverb cloud. Range: 1ms to 350ms
Diffusion Sets the amount of scattering of the late reverb cloud.
Range: 0% to 100%
High Freq. Damping Sets the rate at which high frequencies die away.
Range: -10.0 to +3.0 damping factor
Low Freq. Damping Sets the rate at which low frequencies die away.
Range: -10.0 to +3.0 damping factor
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Vocal Morpher

This unique effect allows you to select two vocal phonemes and morph between them using an LFO. Phonemes are the consonants and vowels we use in articulating speech sounds and these sounds are very distinctive and evocative. 30 different phonemes are available and these can be shifted up or down in pitch for even more effects.
To use the Vocal Morpher, you just select Phoneme A and Phoneme B from the list of thirty. Now the LFO automatically morphs back and forth between the two selected phonemes, creating interesting vocal articulations. The rate of the LFO is adjustable and you can select between Sine, Triangle or Sawtooth waveforms. The sine and triangle waves fade smoothly. The sawtooth wave gradually fades, then jumps abruptly back.
When the frequency of the A or B Phonemes is shifted up or down, entirely new effects can be produced. These frequency controls can also be used to tune the phoneme frequencies to the range of audio you are processing.
Phoneme B
5 - Effects
Core Effects Descriptions
Frequency
Time
Phoneme A
List of Available Phonemes
A E I O U AA
AE AH AO EH ER IH
IY UH UW B D F
G J K L M N
P R S T V Z
Parameter Description
Phoneme A Select any of the available Phonemes for Phoneme A.
Phoneme A Tuning
Adjusts the frequency of Phoneme A up or down 2 octaves in semitone intervals. Range: -24 semitones to +24 semitones
Phoneme B Select any of the available Phonemes for Phoneme B.
Phoneme B Tuning
Adjusts the frequency of Phoneme B up or down 2 octaves in semitone intervals. Range: -24 semitones to +24 semitones
LFO Rate Controls how fast the phonemes morph back and forth.
Range: .01Hz to 10Hz
LFO Waveform Selects the waveform for the morph: Sinusoid, Triangle, Sawtooth
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5 - Effects E-MU PowerFX

E-MU PowerFX

The hardware-accelerated effects of the E-MU Digital Audio System can also be used as VST inserts in Cubase. E-MU PowerFX allow you to use PatchMix DSP effects from within Cubase with minimal load on your CPU.
E-MU PowerFX incorporate smart time alignment technology which automatically compensates for system latencies and ensures proper synchronization of audio throughout the VST chain (if the host application supports this feature).
E-MU PowerFX On/Off Preferences
Input Signal Present
FX Parameters
 E-MU PowerFX are not
available at 88.2kHz, 96kHz, 176.4kHz and 192kHz sample rates.
f Cubase SX/SL 2.0,
Nuendo and Sonar (using the Cakewalk VST adapter 4.4.1) implement VST 2.X auto delay compensation.
FX Palette
FX Inserts
Output Signal Present
FX Presets
Preset Editing
Parameter Description
PowerFX On/Off Enables or bypasses E-MU PowerFX.
FX Palette Select from a single ā€œCoreā€ effect or a Multi -Effect.
FX Inserts Drop Effects from the FX Palette here.
Signal Present LEDs These indicators turn blue to show the presence of input and
output signals.
FX Parameters Select the desired effect in the center insert section, then adjust
the wet/dry mix and parameters for the effect.
FX Presets Select from the list of preprogrammed effect presets here.
Preset Editing Click here to Save, Delete, Rename or Overwrite a User Preset.
See the ā€œUser Preset Sectionā€ for more information
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Parameter Description
Preferences The Preferences menu allows you to:
• Toggle the Tooltips On or Off
• Extra Buffers - Check this box if excessive stuttering occurs when using E-MU PowerFX in your VST Host application. This box should be checked when using Fruity Loops.
• Render Mode - Induces realtime rendering in applications which do not support realtime rendering (WaveLab, SoundĀ­Forge).
5 - Effects
E-MU PowerFX
ī˜‚ To Setup & Use E-MU PowerFX:
Setup Cubase or Cubasis
1. Launch Cubase or Cubasis.
2. Instantiate E-MU PowerFX in an Insert or Aux Send location within Cubase.
3. Press the Insert Edit button in Cubase to bring up the E-MU PowerFX plug-in
window shown on the previous page.
Setup E-MU PowerFX
4. Make sure the blue button is illuminated, indicating that E-MU Power FX is on.
The blue ā€œSignal Presentā€ indicators will be illuminated if E-MU PowerFX is properly patched into a signal path.
5. Drag the desired effects from the Effects Palette to the center Insert strip.
6. Click on the Effect you wish to edit in the center Insert Strip (it will highlight in
yellow), then adjust the effects parameters in the right section of the window.
7. You can also select or edit User Presets from the section below the FX parameters.
See the ā€œ
Add Delay Compensation
If you are using Cubase VST 5.1, you will have to insert an E-Delay Compensator into any other audio tracks to keep them time-aligned.
8. Simply insert an E-Delay Compensator plug-in into the same insert location you
used for E-MU PowerFX on any other audio tracks. That’s it.
User Preset Sectionā€ for more information.
(if needed)
 Using any driver other
than ā€œE-MU ASIOā€ may produce undesirable results when using E-MU PowerFX.
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5 - Effects E-MU PowerFX

Automating E-MU PowerFX

E-MU PowerFX can be automated in Cubase LE (or other recording host) just like any other VST effect. When ā€œWrite Automationā€ is activated in Cubase, control changes made in the PowerFX window during playback will be recorded on a special ā€œAutomation Subtrackā€. When ā€œAutomation Readā€ is activated, the recorded control changes will be played back.
ī˜‚ To Record E-MU PowerFX parameter changes in Cubase LE
1. Add E-MU PowerFX as a Channel Insert.
2. Rewind the song and enable ā€œAutomation Writeā€ by pressing the WRITE button
on, illuminating it. (Refers to Cubase LE. If you are using another application,
refer to the documentation.)
3. Bring the E-MU PowerFX window to the front and select the Effect you want to
automate. The effect parameters appear in the TV screen. Make sure the blue ā€œOnā€ button is lit.
4. Press the Play button on the Cubase Transport control. The song begins playing.
5. Adjust the E-MU PowerFX controls to achieve the effect you want. Rewind the song
when finished.
6. Disable ā€œAutomation Writeā€ and enable ā€œAutomation Readā€ . Playback the song
to hear and view your changes.
7.
To edit Automation, first enable both ā€œAutomation Writeā€ and ā€œAutomation Readā€ and press Play. Cubase LE begins overwriting as soon as you change a control.
8. If you don’t like the results and want to try again, select Show Used Automation
from the Project menu. The Automation Subtrack appears. Next, click in the Parameter Display and select Remove Parameter.
Note: This only erases one automation parameter from the Automation Subtrack. To erase multiple control edits, repeat the procedure above. See the Cubase LE manual for more specific information about automation editing.
 Steinberg Cubasis
does not have the control automation feature.
 Once you have
recorded or drawn automation, do not delete or move effects from the Insert Strip. Doing so will result in unpredictable behavior.

E-MU PowerFX Resource Availability

Because different collections of VST plug-ins and PatchMix Sessions can be run simultaĀ­neously, it is possible to load a Cubase Song or PatchMix Session for which resources are not available. If DSP resources are NOT available for an existing setup:
• E-MU PowerFX loads a Hardware I/O Path and simply passes audio through without any effects. The effects insert slot(s) in E-MU PowerFX will be ā€œredded outā€.
• If no Hardware I/O Paths are available, the plug-in will be disabled and run in a soft pass-through mode. The effects insert slot(s) in E-MU PowerFX will be ā€œgrayed outā€.
• If DSP resources ARE available, but no Hardware I/O Paths are available, the plug-in will run in soft pass-through mode.
• If the sample rate is changed in the middle of a E-MU PowerFX session, E-MU PowerFX plug-ins will be bypassed, since the hardware effects cannot operate at 88kHz, 96kHz, 176.4kHz or 192kHz.
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E-MU PowerFX Compatibility Chart
5 - Effects
E-MU PowerFX
Application Name Compatible? Note Render
Extra
Buffers
Steinberg Cubase VST 5.1 Yes Off Off
Steinberg Cubase SX 1 Yes Off Off
Steinberg Cubase SX 2 Yes Instrument
Off Off Freeze triggers error if not in render mode.
Steinberg Cubase LE Yes Off Off
Steinberg Cubase SL Yes Off Off
Steinberg WaveLab 4 Yes On Off
Steinberg WaveLab Lite (ver 4) Ye s On Off
Steinberg WaveLab 5 No Pops & clicks
On Either may occur. (Try 8 buffers at
1024)
Sony Acid 4 Yes On Off
Sony Vegas 5 Yes On Off
Sony SoundForge 7 No Power FX
On Off crashes when launched.
Adobe Audition 1.5 No Audio
Any Any distortion & immediate lockup.
FruityLoops Studio 4.5 Yes Off On
Ableton Live 3.5 No Distortion
On Off when FX parameters are changed.
Cakewalk Sonar 3 Yes Off Off
E-MU PCI Digital Audio System 81
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5 - Effects Rendering Audio with E-MU PowerFX

Rendering Audio with E-MU PowerFX

Rendering (sometimes called Export) is a mixdown process performed by the host application, which creates a new digital audio file from a multitrack song. Rendering allows a virtually unlimited number of VST effects to be used because the audio processing is performed out of realtime.
E-MU PowerFX and the PatchMix DSP effects are strictly realtime processes. When E-MU PowerFX are used while rendering audio, the rendering process must proceed at realtime rate. Some host applications are not designed to handle realtime rendering and this can cause problems. E-MU PowerFX can be used with these applications if you are willing to follow certain guidelines.

General Tips for Rendering using PowerFX

• If an error message occurs, increase the ā€œASIO Buffer Latencyā€ setting located in the device Setup dialog box. Depending on your setup, you may have to increase or decrease the Buffer Latency settings to find the setting that works.
• Instead of rendering with E-MU PowerFX, bounce the E-MU PowerFX processed tracks to another track in realtime.
• Check ā€œRealtime Renderā€ in the Render dialog box when using Cubase LE, Cubase SX2 or Cubase SL2. This setting will give the best results.

Tips for using Freeze Mode on Cubase LE

•Make the project length as short as possible. Freeze always renders the entire project length, even if the MIDI track being rendered is shorter.
• Great Tip: Temporarily bypass E-MU PowerFX (and any other effects) even when ā€œFreezingā€ another track. This will allow the track to Freeze faster than realtime.

Using E-MU PowerFX with WaveLab and SoundForge

Stuttering in the audio can occur when rendering with SoundForge or any version of Steinberg WaveLab. This problem is caused by discontinuities in the first few audio buffers as they are fed by WaveLab to E-MU PowerFX. The problem can be eliminated by following these guidelines.
• Check ā€œRender Modeā€ box in the E-MU PowerFX preferences. See page 79.
•We recommend that you only use the MME/WAVE E-DSP Wave [xxxx] drivers.
• Reduce the ā€œBuffer Sizeā€ in the WaveLab, Audio Preferences dialog box. This moves the stuttering to beginning of the file.
• Pad the beginning (and/or end) of your audio file with silence (.5 to several seconds depending on the file). This action causes the buffer discontinuities to occur
before
the song begins.
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E-MU VST E-Wire

E-Wire is a special VST/ASIO Bridge which allows you to route digital audio via ASIO to PatchMix and back again.
E-Wire VST incorporates smart time alignment technology that automatically compenĀ­sates for system latencies and ensures proper synchronization of audio throughout the VST chain. In addition, E-Wire also allows you to insert outboard audio gear into the VST environment.
5 - Effects
E-MU VST E-Wire
E-Wire has three main components:
•A VST plug-in which handles the audio routing to PatchMix DSP.
• An ASIO mixer strip in PatchMix DSP configured to route audio to the E-Wire
plug-in. You simply drop the effects you want to use into this strip.
•For hosts that don’t support automatic delay compensation, a manual delay-
compensation plug-in can be inserted in Cubase tracks or channels that don’t use E-Wire to compensate for ASIO delay.
The diagram below may give you a better idea of how E-Wire works:
E-Wire VST plug-in
Send to Strip
E Note: It’s easier to use
E-MU PowerFX instead of E-Wire if you just want to use the hardware effects. (E-Wire was the precursor to E-MU PowerFX.) However, E-Wire can be very useful because it allows you to route VST inserts or Sends to Physical Inputs and Outputs via PatchMix DSP.
Stereo Reverb
Return to VST
ASIO Send
PatchMix DSP
Strip configured
for E-Wire
E-Wire bridges the gap between hardware I/O and the VST world. The E-Wire VST plug-in sends audio to a strip containing the desired effect. An ASIO Send routes the audio back to E-Wire VST.
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5 - Effects E-MU VST E-Wire
ī˜‚ To Setup and use E-Wire:
Setup PatchMix DSP
1. Open PatchMix DSP application.
2. Insert an ASIO Input mixer strip into PatchMix DSP. (Alternately, you can select
ā€œNew Sessionā€, select ā€œE-Wire Exampleā€ and skip to step 6.)
3. Mute the strip or turn the Fader all the way down.
4. Insert an ASIO Send plug-in into one of the inserts on your ASIO strip.
5. Name your ASIO strip as an E-Wire strip.
6. Insert the desired PatchMix DSP effects into slots above the ASIO Send.
7. Save the Session.
Setup Cubase
8. Launch Cubase.
9. Instantiate E-Wire VST in an Insert or Aux Send location within Cubase.
10. Edit the E-Wire plug-in and activate the plug-in by pressing the blue button.
11. Set the ASIO Send and Return on the E-Wire plug-in to match the strip you set up
for E-Wire.
12. Done.
E-Delay Compensation
An E-Delay Compensator must be inserted into any other audio tracks that are not using E-Wire in order to keep them time-aligned.
13. Simply insert an E-Delay Compensator plug-in into the same insert location you
used for E-Wire on any other audio tracks. That’s it.

E-Delay Compensator

As audio is transferred back and forth between the VST host application and the E-MU sound hardware, a delay in the audio stream is incurred. Normally this delay is compenĀ­sated for automatically by the host application, but not all VST host applications support this automatic compensation.
A host will support PowerFX and E-Wire’s plug-in delay compensation if it supports the SetInitialDelay feature of the VST 2.0 specification.
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Currently automatic delay compensation is supported by the Steinberg 2.0 family (Nuendo 2.x, Cubase SX 2.0), Magix Samplitude 7.x, and Sonar (using the Cakewalk VST adapter 4.4.1), but not, unfortunately, by Steinberg Cubase VST 5.1, Cubase LE and Cubasis.
The E-Delay Compensator utility plug-in is used to manually compensate for the transfer delay for hosts that DO NOT support plug-in delay compensation.
The E-Delay Compensator plug-in is used to delay the ā€œdryā€ tracks (tracks without a PowerFX or E-Wire as an insert effect) or auxiliary (send) channels. For each dry track or send, add an E-Delay Compensator plug-in to re-align the track. The E-Delay CompenĀ­sator is automatic and requires no user interaction to operate.
For example, consider a Cubase VST session with two audio tracks. If PowerFX or E-Wire is applied as an insert effect to the first audio track, but not to the second, the first track will be delayed in relation to the second track. The E-Delay Compensator should be added as an insert effect on the second track in order to provide delay compensation.
Cubase VST or Cubasis
5 - Effects
E-MU VST E-Wire
Track 1
Insert
E-Wire
PatchMix
DSP
Track 2
Insert
Track 3
Insert
E-Delay E-Delay
E-Delay Compensator Use
For host applications that don’t support automatic delay compensation.
1.
An E-Delay Compensator should be used when unprocessed audio tracks are played alongside tracks using a PowerFX or E-Wire plug-in.
2. Simply insert an E-Delay Compensator into
each track that doesn’t use a PowerFX or E-Wire send.
E-Delay Units Parameter
The Units value in the E-Delay dialog box should be set for the number of times you send ASIO down to the PatchMix DSP mixer and back in a single track. A single PowerFX insert chain with any number of effects only requires one delay unit because there was only one trip to the hardware and back. If you use two Cubase inserts in series on a track both using PowerFX or E-Wire, you would set the number parameter to 2 on all other audio tracks. Each trip down to PatchMix DSP and back to Cubase equals one unit.
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5 - Effects E-MU VST E-Wire
In practical use, however, you’ll probably never need to use more than one E-Wire VST on a single track since PowerFX effects can be placed in series. We have included this feature ā€œjust in caseā€ you need it.
Here’s one more example of how to use the E-Delay Compensator with different numbers of PowerFX/E-Wire sends on each track. The delay compensation on each track must equal the track with the maximum number of PowerFX/E-Wire sends. See the diagram below.
Cubase VST or Cubasis
Track 1
Insert
PowerFX or E-Wire
Insert
PowerFX
or E-Wire
Track 2
Insert
PowerFX
or E-Wire
Insert
E-Delay
1
Track 3
Insert
E-Delay
2
PatchMix
DSP
Since track 1 uses two PowerFX/E-Wire inserts, the delay of all the other tracks must equal two. Track 2 has one PowerFX/E-Wire insert and so adding one unit of E-Delay keeps it time aligned. Track 3 doesn’t use a PowerFX/E-Wire insert and so it needs two E-Delay Units to remain in alignment.
Grouping Tracks
When several tracks require E-Delay Compensation, you can send the output of each track to a group or bus and use a single E-Delay Compensator on the output of the group or bus.
• E-MU Digital Audio System and PatchMix DSP must be installed.
• E-Wire is compatible with Cubase SX/SL/LE, Cubase VST, Wavelab, and Cakewalk
Sonar (via DirectX-VST adapter) among others.
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6 - Appendix

Using High Sample Rates

Overview

When operating at 176.4k or 192k sample rates, the mixer functionality and number of I/O channels are slightly reduced. The number of ADAT channels also decreases at the 88k/96k and 176/192k sample rates (due to the bandwidth limitations of the optical components).
When using 88.2kHz, 96kHz, 176.4kHz or 196kHz sample rates:
6 - Appendix
Using High Sample Rates
• Effect processors are disabled.
• ADAT is reduced to 4 channels at 88k/96k, and 2 channels at 176k/192kHz.
• ASIO channels are reduced to 8 ASIO (4 stereo) channels at 88k/96k, and 4 ASIO (2 stereo) channels at 176k/192kHz.
• At the 176.4kHz & 192kHz sample rates, the number of physical inputs and outputs is reduced.
• At the 176.4kHz & 192kHz sample rates, S/PDIF optical is disabled
The ADAT optical interface was originally designed to carry 8 channels at a 48kHz sample rate. We use the SonorusĀ® S/MUXā„¢ standard to encode audio with higher sample rates onto the ADAT light pipe. In this multiplexing scheme, two ADAT channels are used to carry one 88.2k or 96k stream and four ADAT channels are used to carry one 176k or 192k audio stream. In order to use the ADAT interface at these higher sample rates, you must have other equipment that supports the Sonorus S/MUX standard.
ī˜‚ Selecting a 176/192k Session
The three possible input configurations are selected by choosing a session template containing the desired I/O from the New Session window. Once you have selected one of the three session types, you will not be able to change to another type without starting a new session.
(Output sends & returns are still available.)
Select the Type of Session you need
Analog & S/PDIF Session
Analog & ADAT Session
S/PDIF & ADAT Session
1. Select New Session from the PatchMix
DSP toolbar.
2. Choose the 176k/192k tab.
3. Select the Template that meets your
requirements and click OK.
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6 - Appendix Using High Sample Rates
&
&
or...
or...
&
At the 192kHz sample rate, you may choose one of these three options:
1. Keep all Analog I/O, but lose S/PDIF 3. Keep S/PDIF & ADAT, but lose
2. Keep all Analog I/O, but lose ADAT Line Inputs 2L/2R & Line Outputs 3L/3R
E-MU 1616 Hardware Inputs & Outputs at 176.4k or 192k
Source Input
Analog &
SPDIF
ADAT 0 0 2 2 2 2
Microphone 2 - - - - - - - - 2 - - - - - - - - 2 - - - - - - - -
Line 1 2 2 2 2 2 2
Line 2 2 2 0 2 2 2
Line 3
(output)
S/PDIF 2 2 2 2 0 0
Total
- - - - - - - - 2 - - - - - - - - 2 - - - - - - - - 2
8 8 8 8 8 8
Output
Analog &
SPDIF
Input
SPDIF &
ADAT
Output
SPDIF &
ADAT
Input
Analog &
ADAT
Output
Analog &
ADAT
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WDM Recording and Playback Behavior

WDM recording and playback is supported at all PatchMix sample rates. The behavior of the driver with respect to PatchMix sample rate is described below.
When PatchMix and the WDM audio content (.WAV file format, playback and record settings in WaveLab. etc.) are both running at the same sample rate, and when a Wave strip or send is present in the PatchMix mixer configuration, WDM audio will be played or recorded ā€œbit accurateā€ without sample rate conversion or bit truncation.
When running PatchMix at 44kHz/48kHz, if there is a mismatch between the WDM playback audio content and the PatchMix sample rate, sample rate conversion is performed, so that WDM audio will always be heard or recorded. Also, such non-nativeĀ­sample-rate audio is truncated to 16-bits.
When running PatchMix at the higher sample rates of 88.2kHz, 96kHz, 176.4kHz or 192kHz, WDM record or playback audio content must be running at the same sample rate as PatchMix. If the sample rates are mismatched, NO AUDIO will be produced or recorded. In other words, the WDM driver does not perform sample rate conversion of any kind when PatchMix is running at 88.2kHz, 96kHz, 176.4kHz or 192kHz.
6 - Appendix
Using High Sample Rates
E-MU PCI Digital Audio Systems 89
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6 - Appendix Sync Daughter Card Supplement

Sync Daughter Card Supplement

SMPTE Conversion

One of the main functions of the Sync Daughter Card is to convert SMPTE (LTC) to MIDI Time Code (MTC) and vice-versa. The term ā€œHost MTCā€ refers to MTC, which is generated or used by the host application (Cubase, etc.). MTC is also available at the MIDI jack on the back of the Sync Card.
SMPTE Features
• Conversion of SMPTE to MTC quarter-frame messages & full-frame messages.
Constant quarter-frame messages are generated with steady SMPTE data input. Occasional MIDI full-frame messages are generated when SMPTE contains data dropouts.
• Conversion of MTC (quarter frame & full frame) messages from the host
computer to SMPTE out.
Simultaneous SMPTE and MTC output when receiving MTC from the host computer
• Outputs SMPTE and MTC striping data.
SMPTE Start Time and type can be set via System Settings dialog box.
 Warning: SMPTE and
MTC do not provide sample sync for digital I/O. You must use Word Clock, S/PDIF or ADAT sync.

SMPTE Options

When the Sync Daughter Card is installed in your system a SMPTE button in the PatchMix DSP mixer becomes visible. Pressing the SMPTE button brings up the SMPTE window.
Current Time
SMPTE Status/
Error Indicators
Start Striping
SMPTE Frame
Rate Settings
SMPTE
Start Time
Flywheel Mode
• Off • Continuous
• Fixed •1-time Jam
Stop
Striping
Output Source
Flywheel
Amount
Word Clock Termination
Output
Level
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6 - Appendix
Sync Daughter Card Supplement
Mode (fps)
SMPTE Striping
Stripe Button
Stop Button
Flywheel Mode
Output Level
FLY/JAM Frames
Word Clock Termination
SMPTE/MTC Output Source
Sets the transmitted frame rate when striping SMPTE.
Edit this field to set the start time in hours:minutes:seconds:frames for striping SMPTE.
Initiates SMPTE Time Code generation at the SMPTE output beginning at the time set in the striping display.
Stops SMPTE striping. This button also stops SMPTE when One-Time Jam Sync has been initiated.
Selects one of the four Flywheel modes. See the descriptions below.
Sets the SMPTE output level from -10dBV (consumer) to +4dBu (pro).
If flywheel mode is on and a dropout is detected this is the number of flyĀ­wheel frames that will be output before the sync card stops and chases.
Turns word clock termination on or off. Except in special cases, this conĀ­trol should normally be left on. See ā€œ
This control selects the source of the SMPTE output jack. The choices are: Host MTC or the SMPTE Input jack (to regenerate SMPTE).
Word Clock In/Outā€.

SMPTE Modes of Operation

Host Mode
The host computer is the source of synchronization. MTC messages are sent to the Sync Daughter Card from the computer application and converted into SMPTE. MTC is also output from the MIDI port on the Sync Daughter Card.
External Mode
SMPTE messages from SMPTE In are converted to MTC (quarter-frame messages) and sent to the host application. This happens automatically whenever LTC is received at the SMPTE input jack. Clean SMPTE data is also transmitted from SMPTE Out if ā€œSMPTE (Regenerate)ā€ is set.
Flywheel Mode
If the incoming SMPTE data is corrupted or missing frames, MTC code will continue to be output if ā€œFlywheel modeā€ is enabled. The flywheel modes are described below.
Flywheel Modes
Off
Fixed 0-127
Continuous
1-Time Jam Sync
Upon any dropout, MTC stops and the Sync card monitors the input for valid code. If valid code is again received, it chases and relocks.
Upon any dropout, MTC continues outputting Quarter-frame messages at the same rate (flywheeling). When a dropout is detected, this is the number of frames that will be output before the Sync card stops outputĀ­ting MTC and monitors the input for valid code. If valid code is again received, it chases and relocks.
Upon any dropout, MTC continues outputting Quarter-frame messages at the same rate (flywheeling). The Sync card monitors the input for valid code and continues flywheeling until valid code is received, then relocks.
Upon any dropout, MTC continues outputting Quarter-frame messages at the same rate (flywheeling) without monitoring the SMPTE input until the Stop button is pressed.
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6 - Appendix
 When the MicroDock
is connected to the 1010 PCI card, the digital I/O on the PCI card is disabled. Use the digital I/O on the MicroDock.
A
Sync Daughter Card Supplement
Stripe Mode
This mode is used to record SMPTE time code onto an audio track of another recorder. SMPTE is output when the Start button is pressed in the System Settings menu and begins at the time set by the Start Time setting. MTC is also simultaneously output from the Sync Daughter Card MIDI out. SMPTE and MTC will continue to be output until the Stop button is pressed.

Example SMPTE Connection

In the diagram below, Cubase is controlling the entire system by sending MTC to the Sync Card which converts MTC to SMPTE. SMPTE is fed to the ADAT/BRC to convey the absolute time information (hours-minutes-seconds-frames). ADAT/BRC is the word clock master, controlling the Digital Audio System either through the embedded clock in the ADAT optical stream or using word clock.
Cubase
1010
CARD
EDI
DAT
In
SYNC
CARD
optional
if ADAT
sync isn't
used
LTC
MTC
BRC
1
2
33
34
65
66
97
98
9
10
41
42
73
74
105
106
17
18
49
50
81
82
113
114
25
26
57
58
89
90
121
122
TRACK 1-32 TRACK 33-64 TRACK 65-96 TRACK 97-128
AUTO INPUT ALL SAFE ALL INPUT ALL CLEAR
GROUP 1 GROUP 2 GROUP 3 GROUP 4 SET GROUP
MASTER REMOTE CONTROL
7
39
71
103
15
47
79
111
23
55
87
119
31
63
95
127
8
40
72
104
16
48
80
112
24
56
88
120
32
64
96
128
PITCH UP
PITCH DOWN
PITCH MODE
MIDI UTIL
COPY SONGDELETE SONG
SET LOCATELOCATE SONG
PITCH CONTROL TAPE LOCATION
CENTS
HOURS MINUTESSECONDS FRAMES
PITCH MODE
FIXED VARIABLE
SMPTE IN MIDI IN
NAME
TEMPO MAP
RECORD XFADE
7
STUV8WXYZ0(CHARS)
SAVE SETUP
TO TAPE
4
JKL5MNO6PQR
LOAD SETUP
1
FROM TAPE
ABC2DEF3GHI
LOCATE 0
SET LOCATE
LOCATE
BARS BEATS SUB BEATS
CURSOR
RECORD
INPUT
3
4
5
6
35
36
37
38
67
68
69
70
99
100
101
102
RECORD
INPUT
11
12
13
14
43
44
45
46
75
76
77
78
107
108
109
110
RECORD
INPUT
19
20
21
22
51
52
53
54
83
84
85
86
115
116
117
118
RECORD
INPUT
27
28
29
30
59
60
61
62
91
92
93
94
123
124
125
126
NORMAL
ABSOLUTE
RELATIVEE
DROP FRAME
29.97 FPS
SMPTE
BARS
30 FPS
25 FPS
24 FPS
COPY TAPE LOCATION
AUTO-PUNCH
DIGITAL I/O
AUTO PLAY
DISPLAY TYPE EJECT
DISPLAY MODE RESET 0 FORMAT TAPE
SMPTE START
SMPTE
PRE-ROLL POST-ROLL
LOOP TAPE OFFSETTRACK DELAY
GEN SYNC EDIT
REHEARSE
RECORDPLAYSTOPFAST FORWARDREWIND
EXT SYNC
OFFSET
PatchMix DSP set to
The Sync Card should not be used as both the SMPTE and word clock master. Word Clock is generated by the Digital Audio System and NOT by the software application (Cubase). SMPTE is not locked to Word Clock inside the Sync Card—they are completely independent.
92 Creative Professional
ADAT Sync
ADAT Optical
carries embedded
word clock
ADAT
ADAT 9-pin
REWIND FAST FWD STOP PLAY RECORD
EJECT
Page 93

MIDI Time Code (MTC)

MIDI time code is basically SMPTE time code adapted to the world of MIDI. MTC specifies ā€œabsoluteā€ location information in hours:minutes:seconds:frames, just like SMPTE. There are two main kinds of messages in MTC: Full-frame messages and Quarter-frame messages.
Full-frame messages are ten bytes long and are sent when SMPTE start, stops, or relocates. Full-frame messages contain the entire SMPTE number of, hours, minutes, seconds, frames, as well as the SMPTE type: 24fps, 25fps, 30fps non-drop, 30fps drop.
Quarter-frame messages are sent at each quarter of a SMPTE frame and only carry 1/8th of the SMPTE time message. Quarter-frame messages require two entire SMPTE frames to send the complete time stamp (h:m:s:f). Timing accuracy is maintained as long as the quarter-frame messages continue to come in at a constant rate.
ī˜‚ To Enable MTC:
MIDI Time code disables the use of MIDI port 2 on the MicroDock
1. Open Session Settings from the toolbar.
2. Select the MIDI tab and choose Sync Card/MTC from the MIDI options.
3. Click OK to close the window.
Since it is important to have a stable timing reference for your song or sequence, we have given MTC its own MIDI output port on the Sync Daughter Card. This ensures that the timing information will not be affected by other MIDI data on the line.
6 - Appendix
MIDI Time Code (MTC)
 MTC and SMPTE do
NOT synchronize at the sample rate and are not locked to word clock in any way.
SMPTE and MTC are used to synchronize music but do not have the required resolution to sample-lock digital audio.

Word Clock In/Out

Word clock provides a standardized means of synchronizing multiple digital audio devices so that data can be transferred digitally. In order to digitally transfer from one device to another, the two devices MUST be synchronized. Clicks and pops in the audio will result when transferring digital audio which is not synchronized.
The E-MU 1010 PCI card can be externally clocked from the ADAT input, S/PDIF input or from the Sync Daughter card (if installed). In a digital studio, all digital devices in the system should run off the same master Word Clock.
ī˜‚ To Synchronize PatchMix DSP to an External Clock Source:
1. Make sure an external clock source is connected to the E-MU Digital Audio System
hardware via the word clock, ADAT or S/PDIF input.
2. Open the Session Settings dialog box.
3. Under the System tab, select External Source, then select either word clock, ADAT
or S/PDIF.
4. Press OK to close the dialog box.
5. Check the Sync section of PatchMix DSP to verify that the Locked indicator is
illuminated.
Devices can be connected in daisy chain fashion (word clock out connected to the next unit’s word clock in) or in parallel for one or two devices, but professional digital studios normally use a master word clock generator or ā€œHouse Syncā€ with a distribution system so that each device receives a phase-coherent and jitter-free word clock.
f Word clock, ADAT and
S/PDIF synchronize at the sample rate and are used to transfer digital data between machines.
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6 - Appendix
A
Word Clock In/Out
Digital
Device 1
Digital
Device 3
House Sync
Generator
Digital
Device 2
A master word clock generator is preferable for larger digital setups.
Word Clock In: Receives word clock (sample clock) from another digital device such as a digital video deck, digital recorder or digital mixer.
Word Clock Out: Sends word clock (sample clock) to another digital recorder. Word clock is always output, whether it is generated by the internal clock or passed through from the word clock input.
75Ω On/Off: Termination for the word clock input can be switched on or off in the Sync Card menu of the PatchMix DSP application. Normally word clock termination should be left on. If you have problems with a weak word clock signal, try turning termiĀ­nation off. See Word Clock Termination.
The diagram below shows the proper way to connect and terminate a serial word clock chain. Using a BNC ā€œTā€ connector ensures that word clock is precisely in phase for both devices. The middle device has termination turned Off and the last device in the word clock chain has termination turned On.
Digital
Device 4
Digital Mixer
 When the MicroDock
is connected to the 1010
Word Clock
DAT Optical
or AES Digital
ADAT Optical
Word Clock Termination OFF
T - connector
IN
Word Clock
PCI card, the digital I/O on the PCI card is disabled. Use the digital I/O on the MicroDock.
AES
E-MU 1010 CARD
EDI
IN
SYNC CARD
Word Clock Termination ON
This diagram shows the proper way to connect word clock if you don’t have a multi-output word clock generator. The last device in a Word Clock chain should have Termination ON.
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Page 95

Getting in Sync

 When the MicroDock
is connected to the 1010 PCI card, the digital I/O on the PCI card is disabled. Use the digital I/O on the MicroDock.
Whenever you connect external digital audio devices together, you need to be aware of how they are synchronized to each other. Simply connecting digital out to digital in doesn’t guarantee that two digital devices are synced, even if audio is being passed. Unless you have set one to be the Master and the other a Slave, they are probably NOT synchronized and the quality of your audio will suffer.
S/PDIF and ADAT are two commonly used digital audio formats. Both these digital formats carry an embedded word clock which can be used to synchronize the digital equipment. You must enable ā€œExternal Clockā€ on the slave device to have clock sync!
The diagrams below show two ways to synchronize an external A/D - D/A converter to the E-MU Digital Audio System using the ADAT lightpipe connection.
In the first example, only the A/D converters on the external device are being used. Only one lightpipe is needed as long as PatchMix is set to receive its word clock signal from the external device. The external A/D is the Master and the E-MU DAS is the Slave.
6 - Appendix
Getting in Sync
EXTERNAL
Slave
EXTERNAL
External Device supplies Master Clock
The lightpipe carries eight
channels of audio data and
an embedded clock.
1 2 3 4 5 6 7 8
Master
Set PatchMix DSP to receive:
External ADAT Sync
PatchMix DSP supplies Master Clock
This lightpipe carries eight
channels of audio data.
(via ADAT)
ADAT Out
External A/D - D/A Converter
(via ADAT)
Set External Device to receive:
ADAT Out
External ADAT Sync
Master
In the second example a second lightpipe is used to supply ā€œembedded word clockā€, as well as eight channels of audio to the external A/D - D/A. The external device MUST be set to receive external clock via ADAT or the units will not be synchronized. The E-MU Digital Audio System is the Master and the external A/D - D/A is the Slave.
E-MU PCI Digital Audio Systems 95
Slave
1 2 3 4 5 6 7 8
ADAT In
This lightpipe carries an
embedded clock signal
& eight channels of audio.
External A/D - D/A Converter
Page 96
6 - Appendix Useful Information

Useful Information

Cables - balanced or unbalanced?

All inputs and outputs on the E-MU Digital Audio System are designed to use either balanced or unbalanced cables. Balanced signals provide an additional +6dB of gain
on the inputs and are recommended for best audio performance, although unbalanced cables are fine for most applications. If you’re having problems with hum and noise or just want the best possible performance, use balanced cables.
Balanced Cables
Balanced cables are used in professional studios because they cancel out noise and interference. Connector plugs used on balanced cables are XLR (3-prong mic connector) or TRS (Tip, Ring, Sleeve) 1/4" phone plugs.
Balanced XLR Connectors
21
3
12
3
1 = Ground/shield 2 =
Hot (+)
3 =
Cold (-)
OutputInput
Sleeve = Ground
Balanced 1/4ā€ TRS Connectors
Sleeve = Ground
Unbalanced 1/4ā€ Connectors
Tip = Hot (+)
Ring = Cold (-)
Tip = Signal
 WARNING: Do NOT
use balanced audio cables when connecting balanced outputs to unbalanced inputs. Doing so can increase noise level and introduce hum. Use balanced (3-conductor) cables ONLY if you are connecting balanced inputs to balanced outputs.
Balanced cables have one ground (shield) connection and two signal-carrying conductors of equal potential but opposite polarity. There is one ā€œhotā€ or positive lead, and a ā€œcoldā€ or negative lead. At any point in time, both conductors are equal in voltage but opposite in polarity. Both leads may pick up interference, but because it is present both in and out of phase, this interference cancels out at the balanced input connection.
Unbalanced Cables
Unbalanced cables have one conductor and one ground (shield) and usually connect via unbalanced 1/4" phone plugs or RCA phono plugs. The shield stays at a constant ground potential while the signal in the center conductor varies in positive and negative voltage. The shield completely surrounds the center ā€œhotā€ conductor and is connected to ground in order to intercept most of the electrical interference encountered by the cable. Unbalanced cables are more prone to hum and interference than balanced cables, but the shorter the cable, the less hum and noise is introduced into the system.
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Adapter Cables

r
1/8ā€ Mini-phone to 1/4ā€ Adapters
To connect headphones with an 1/8ā€ (mini-phone) plug to the headphone jack on the MicroDock, you need a 1/8ā€ to 1/4ā€ adapter. These handy devices are available at electronic department stores everywhere.
1/8" to 1/4"
Headphones with 1/8" plug
Cinch (RCA) to 1/4ā€ Adapters
Equipment (such as consumer audio gear) which uses Cinch/RCA type connectors can be connected to the MicroDock using readily available adapter cables. These adapters can be found at most stores that sell audio equipment.
Headphone Adapte
6 - Appendix
Useful Information
Tip = Hot (+)
Cinch/RCA
Plug
Sleeve = Ground
1/4" Phone Plug
Shaft = Ground
Tip = Hot (+)

Digital Cables

Don’t cheap out! Use high quality optical fiber Toslink (ADAT) cables. It’s also a good idea to keep digital cabling as short as possible (1.5 meters for plastic light pipes; 5 meters for high quality glass fiber light pipes).
Use low-capacitance, video-grade cable for coaxial S/PDIF to avoid data corruption.

AES/EBU to S/PDIF Cable Adapter

This simple adapter cable allows you to receive AES/EBU digital audio via the S/PDIF input on the E-MU 1010 PCI card. This cable may also work to connect S/PDIF out from the 1010 PCI card to the AES/EBU input of other digital equipment.
From AES/EBU
Device
To S/PDIF
In
N.C.
12
+
3
-
E-MU PCI Digital Audio Systems 97
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6 - Appendix
V
Useful Information

Grounding

In order to obtain best results and lowest noise levels, make sure that your computer and any external audio devices are grounded to the same reference. This usually means that you should be using grounded AC cables on both systems and make sure that both systems are connected to the same grounded outlet. Failure to observe this common practice can result in a ground loop. 60 cycle hum in the audio signal is almost always caused by a ground loop.

Phantom Power

Phantom power is a dc voltage (+48 volts) which is normally used to power the pream­plifier of a condenser microphone. Some direct boxes also use phantom power.
Pins 2 and 3 of the MicroDock microphone inputs each carry +48 volts dc referenced to pin 1. Pins 2 and 3 also carry the audio signal which ā€œridesā€ on top of the constant 48 volts DC. Coupling capacitors at the input of the MicroDock block the +48 volt DC component before the signal is converted into digital form. The audio mutes for a second when phantom power is turned on.
After turning phantom power off, wait two full minutes before recording to allow the DC bias to drain from the coupling capacitors since the bias could affect the audio headroom.
Balanced dynamic microphones are not affected by phantom
1
(grd)
3
Since ribbon microphones are fairly specialized and generally expensive, you’ll know if you own one. Most microphones are either of dynamic or condenser type and these are not harmed by phantom power.
2
+48
power. An unbalanced dynamic microphone may not work properly, but will probably not be damaged if phantom power is left on.
Ribbon microphones should NOT be used with phantom power on. Doing so can seriously damage the ribbon element.

Appearance Settings in Windows

Adjusting the ā€œPerformance Optionsā€ in Windows will improve the screen appearance when moving the mixer around on the screen.
ī˜‚ To Improve the Appearance Settings:
1. Open the Windows Control Panel. (Start, Settings, Control Panel).
2. Select System. Select the Advanced Settings tab.
3. Under Visual Effects, select Adjust for Best Performance. Click OK.
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Technical Specifications
Specifications: 1616m PCI System
GENERAL
6 - Appendix
Technical Specifications
Sample Rates
Bit Depth
Hardware DSP
Converters & OpAmps
WDM Drivers
MicroDockm Power Use
ANALOG LINE INPUTS
Type
Level (software selectable)
Frequency Response
THD + N
SNR
Dynamic Range
Channel Crosstalk
Common-mode Rejection
Input Impedance
44.1 kHz. 48 kHz, 96 kHz, 192 kHz from internal crystal Accepts externally supplied clock from S/PDIF or ADAT
16 or 24-bits
100MIPs custom audio DSP. Zero-latency direct hardware monitoring with effects
ADC - AK5394 (AKM) DAC - CS4398 (Cirrus Logic) OpAmp - NJM2068M, NJM2122, NJM2082 (JRC)
2-in/8-out at 44.1kHz, 48kHz, 88.2kHz, 96kHz 2-in/4-out at 176.4kHz & 192kHz
.32A @ +48VDC 15.4 Watts
Servo-balanced, DC-coupled, low-noise input circuitry
Professional: 18 dBV maximum (balanced)
Consumer: 6 dBV maximum (unbalanced)
20 Hz - 20 kHz, +0.0/-0.03 db
-110 dB (.0003%) (-1 dBFS, 20kHz BW)
120 dB (A-weighted, 20kHz BW)
120 dB (A-weighted, 20kHz BW)
-120 dB, (1 kHz)
-79 dB at 60Hz
10K ohm
ANALOG LINE OUTPUTS
Type
Level (software selectable)
Frequency Response
THD + N
SNR
Dynamic Range
Stereo Crosstalk
Output Impedance
E-MU PCI Digital Audio Systems 99
Balanced, low-noise, 2-pole low-pass differential filter, AC-coupled
Professional: 18 dBV maximum (balanced)
Consumer: 6 dBV maximum (unbalanced)
20 Hz - 20 kHz, +0.0/-0.03 dB
-105 dB (.0005%) (-1dBFS, 20kHz BW)
120 dB (A-weighted, 20kHz BW)
120 dB (A-weighted, 20kHz BW)
< -115 dB, 1kHz
560 ohms
Page 100
6 - Appendix Technical Specifications
Specifications: 1616m PCI System
MIC PREAMP/LINE INPUT
HI-Z LINE INPUT
Gain Range:
Max Level:
THD+N:
SNR:
Dynamic Range:
Crosstalk:
Frequency Response:
Input Impedance:
CMRR:
-15 to +51 dB
19 dbV (21.2 dBu)
-105 dB (.00056%), -1 dBFS, 20kHz BW
118 dB, (A-weighted, 20kHz BW)
118 dB, (A-weighted, 20kHz BW)
Minimum Gain: -140 dB (1kHz) Maximum Gain: -80 dB (1kHz)
+0.10/-0.00 dB, 20 Hz - 20 kHz
1M ohm
-44 dB (60Hz)
MICROPHONE PREAMP
Gain Range:
Max Level:
THD+N:
SNR:
Frequency Response:
Input Impedance:
CMRR:
Crosstalk:
HEADPHONES
Frequency Response:
THD+N: (1 kHz, max. level)
SNR:
Dynamic Range:
Stereo Crosstalk:
Max Output Power:
Output Impedance:
-1.3 to +65 dB
-5.6 dbV (7.8 dBu)
Min Gain: -112dB (.00025%), (1kHz, -1 dBFS, 20kHz BW) 40dB Gain: -96dB (.0015%) 60dB Gain: -75dB (.017%)
119 dB (A-weighted, min. gain)
20 Hz - 20 kHz ±0.08 dB (20Hz - 20kHz, gain +40dB)
1.5 Kohm
-95 dB (60Hz, 35 dB gain, -1 dBFS)
-100 dB (1 kHz, 40 dB gain, -1 dBFS)
+0.0/-0.07 dB, 20 Hz - 20 kHz
24 ohm load: -85 dB (0.018%) 600 ohm load: -94 dB (0.002%)
118 dB (A-weighted)
118.5 dB (A-weighted)
24 ohm load: < -43 dB (1kHz at 0 dBFS) 600 ohm load: < -100 dB (1kHz at 0 dBFS)
100 mW (24Ω load)
22 ohms
Gain Range:
100 Creative Professional
85 dB
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