Creative | 1616m | Owner`s manual | Creative 1616m Owner`s manual

Digital Audio System
Owner's Manual
E-MU 1616/1616M CardBus Digital Audio System
1
E-MU 1616/1616m CardBus
Digital Audio System
Owner’s Manual
© 2003 E-MU Systems
All Rights Reserved
Software Version: 1.8
E-MU World Headquarters
Europe, Africa, Middle East
E-MU Systems
Creative Labs
1500 Green Hills Road
Ballycoolin Business Park
Scotts Valley, CA USA
Blanchardstown
95067-0015
Dublin 15
IRELAND
2
E-MU Japan
Creative Media K K
Kanda Eight Bldg., 3F
4-6-7 Soto-Kanda
Chiyoda-ku, Tokyo 101-0021
JAPAN
www.japan.creative.com
Creative Professional
Table of Contents
1- Introduction ................................................................. 7
Welcome!.............................................................................................................................. 7
The E-MU 02 CardBus Card ............................................................................................ 7
E-MU MicroDock ............................................................................................................. 7
E-MU 1616M System ....................................................................................................... 8
PatchMIx DSP ................................................................................................................... 8
Notes, Tips and Warnings ............................................................................................ 8
2 - Installation .................................................................. 9
Setting up the 1616 or 1616m system ................................................................................ 9
Notes for Installation ................................................................................................... 9
Installing the CardBus Card and Software........................................................................ 10
Plug in the E-MU 02 CardBus Card .............................................................................. 10
Software Installation .......................................................................................................... 10
Installing the E-MU 02 Drivers .................................................................................. 10
Windows 2000 or Windows XP ................................................................................ 10
Uninstalling all Audio Drivers and Applications ..................................................... 10
Note About Windows Logo Testing .......................................................................... 11
Connecting the MicroDock ............................................................................................... 11
Connector Types ............................................................................................................ 11
3 - CardBus Card & MicroDock ......................................... 13
The E-MU 02 CardBus Card .............................................................................................. 13
Connections ................................................................................................................... 13
CardBus Connector .................................................................................................... 13
Removing the CardBus Card ..................................................................................... 13
EDI Connector ............................................................................................................ 13
Monitor Output .......................................................................................................... 13
The MicroDock................................................................................................................... 14
Front Panel Connections ............................................................................................... 15
Preamp Section ........................................................................................................... 15
S/PDIF Digital Audio Input & Output ...................................................................... 15
ADAT Optical Digital Input & Output ...................................................................... 16
Headphone Output & Volume Control .................................................................... 16
Rear Panel Connections ................................................................................................. 18
Line Level Analog Inputs ........................................................................................... 18
Turntable Inputs & Ground Lug ................................................................................ 18
Line Level Analog Outputs ........................................................................................ 18
Computer Speaker Analog Outputs .......................................................................... 19
MIDI 1 & 2 In/Outs .................................................................................................... 19
EDI Connector (Card) ............................................................................................... 19
E-MU 1616/1616M CardBus Digital Audio System
3
4 - The PatchMix DSP Mixer ............................................. 21
PatchMix DSP..................................................................................................................... 21
Overview of the Mixer........................................................................................................ 21
Mixer Window ................................................................................................................ 22
Mixer Block Diagram ..................................................................................................... 22
Pre Fader or Post Fader .............................................................................................. 22
E-MU Icon in the Windows Taskbar ................................................................................. 23
The Toolbar ........................................................................................................................ 23
The Session ......................................................................................................................... 24
New Session .................................................................................................................... 24
Open Session .................................................................................................................. 25
Save Session .................................................................................................................... 25
Session Settings .............................................................................................................. 25
System Settings ........................................................................................................... 25
Using External Clock .................................................................................................. 26
I/O Settings ................................................................................................................. 26
Input Mixer Strips............................................................................................................... 28
Input Type ................................................................................................................... 28
Mixer Strip Creation........................................................................................................... 29
Insert Section .................................................................................................................. 30
Working with Inserts .................................................................................................. 30
The Insert Menu ......................................................................................................... 31
ASIO Direct Monitor Send/Return ............................................................................ 32
Meter Inserts ............................................................................................................... 33
To Set the Input Levels of a Strip ................................................................................... 34
Making the Best Possible Recording ......................................................................... 34
Trim Pot Insert ............................................................................................................ 35
Test Tone/Signal Generator Insert ............................................................................. 36
Managing Your Inserts ................................................................................................... 37
Aux Section ..................................................................................................................... 38
Sidechain Diagram ..................................................................................................... 38
Pre or Post Fader Aux Sends ...................................................................................... 39
Level, Pan, Solo & Mute Controls ................................................................................. 40
Main Section....................................................................................................................... 41
TV Screen & Selectors ..................................................................................................... 42
Effect ........................................................................................................................... 42
Input ........................................................................................................................... 43
Output ........................................................................................................................ 43
Auxiliary Effects & Returns ............................................................................................ 44
Sidechain Diagram ..................................................................................................... 44
Sync/Sample Rate Indicators ......................................................................................... 44
Output Section ............................................................................................................... 45
Main Inserts ................................................................................................................ 45
Main Output Fader ..................................................................................................... 45
Output Level Meters ................................................................................................... 45
Monitor Output Level ................................................................................................ 45
Monitor Balance Control ........................................................................................... 45
Monitor Output Mute ................................................................................................ 45
4
Creative Professional
5 - Effects ....................................................................... 47
Overview............................................................................................................................. 47
The Effects Palette............................................................................................................... 47
FX Insert Chains ............................................................................................................. 48
Creating, Renaming & Deleting Categories or Presets ............................................. 49
FX Edit Screen..................................................................................................................... 50
User Preset Section ......................................................................................................... 51
Core Effects and Effects Presets ..................................................................................... 52
List of Core Effects.............................................................................................................. 53
DSP Resource Usage ....................................................................................................... 53
Core Effects Descriptions................................................................................................... 54
1-Band Para EQ .............................................................................................................. 54
1-Band Shelf EQ ............................................................................................................. 54
3-Band EQ ...................................................................................................................... 55
4-Band EQ ...................................................................................................................... 56
Auto-Wah ........................................................................................................................ 57
Chorus ............................................................................................................................ 58
Compressor .................................................................................................................... 58
Basic Controls ............................................................................................................. 59
Distortion ....................................................................................................................... 60
Flanger ............................................................................................................................ 61
Freq Shifter ..................................................................................................................... 62
Leveling Amp .................................................................................................................. 63
Lite Reverb ...................................................................................................................... 64
Mono Delays - 100, 250, 500, 750, 1500, 3000 .......................................................... 65
Phase Shifter ................................................................................................................... 66
Rotary .............................................................................................................................. 66
Speaker Simulator .......................................................................................................... 67
Stereo Delays - 100, 250, 500, 750, 1500 .................................................................... 68
Vocal Morpher ................................................................................................................ 70
E-MU PowerFX ................................................................................................................... 71
Automating E-MU PowerFX .......................................................................................... 73
E-MU PowerFX Resource Availability ........................................................................... 73
Rendering Audio with E-MU PowerFX ............................................................................. 75
General Tips for Rendering using PowerFX .............................................................. 75
Tips for using Freeze Mode on Cubase LE ................................................................ 75
Using E-MU PowerFX with WaveLab and SoundForge ............................................... 75
E-MU VST E-Wire ............................................................................................................... 76
E-Delay Compensator .................................................................................................... 77
E-Delay Compensator Use ......................................................................................... 78
E-Delay Units Parameter ............................................................................................ 78
Grouping Tracks ......................................................................................................... 79
6 - Using High Sample Rates ........................................... 81
Overview............................................................................................................................. 81
E-MU 1616 System at 176kHz or 192kHz ................................................................... 81
WDM Recording and Playback Behavior ...................................................................... 83
E-MU 1616/1616M CardBus Digital Audio System
5
7 - Appendix ................................................................... 85
Useful Information ............................................................................................................ 85
Cables - balanced or unbalanced? ................................................................................ 85
Balanced Cables ......................................................................................................... 85
Unbalanced Cables .................................................................................................... 85
Adapter Cables ............................................................................................................... 86
1/8” Mini-phone to 1/4” Adapters ............................................................................ 86
Cinch (RCA) to 1/4” Adapters ................................................................................... 86
Digital Cables ................................................................................................................. 86
AES/EBU to S/PDIF Cable Adapter ............................................................................... 86
Grounding ...................................................................................................................... 87
Phantom Power .............................................................................................................. 87
Appearance Settings in Windows .............................................................................. 87
Technical Specifications..................................................................................................... 88
Internet References............................................................................................................. 95
Forums ........................................................................................................................ 95
Index .............................................................................. 99
6
Creative Professional
1- Introduction
Welcome!
1- Introduction
Welcome!
Thank you for purchasing the E-MU 1616 or E-MU 1616m CardBus 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 multi-channel recording
and playback to any CardBus equipped PC.
1616 & 1616M System Components
E-MU 1616 & 1616m
• E-MU 02 CardBus Card
• E-MU MicroDock
• EDI (E-MU Digital Interface Cable)
• E-MU 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 (with +48V phantom power)
(2) Turntable Preamp Inputs (with RIAA equalized preamplifier)
(1) Stereo Headphone Output (with volume control)
(3) Stereo Computer Speaker Outputs (with 1/8” jacks to connect powered speakers)
The E-MU 02 CardBus Card
The E-MU 02 CardBus Card is the heart of both 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 02 CardBus Card has its own 24-bit stereo
output and can be used without the E-MU MicroDock to drive headphones or line level
inputs.
E-MU MicroDock
Both systems include the E-MU MicroDock, which is a half rack-space, audio interface.
The MicroDock adds the following input and output capabilities: two mic/line inputs
with pro studio-class microphone preamps, 4 balanced line level analog inputs, an RIAA
stereo turntable preamp, 6 balanced line level outputs, a headphone output with front
panel volume control , two sets of MIDI I/O ports, eight-channels of ADAT® optical
digital input and output, as well as a S/PDIF stereo digital input and output. In
addition, three stereo mini phone jacks allow easy connection to powered speaker
systems. You have a total of 16 inputs and 16 outputs! High-quality, 24-bit A/D and
D/A converters are used throughout.
E-MU 1616/1616m CardBus Digital Audio System
7
1- Introduction
Welcome!
E-MU 1616M System
The E-MU 1616m system includes the MicroDockM, and is a no compromise,
mastering-grade system, which includes all the features of the 1616 system. The 1616M
system is distinguished by the addition of ultra-high performance 24-bit/192kHz
A/D - D/A converters which deliver an unbelievable 120dB dynamic range.
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 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.
8
Warnings are especially important, since they help you avoid activities that can
cause damage to your files, your computer or yourself.
Creative Professional
2 - Installation
Setting up the 1616 or 1616m system
2 - Installation
Setting up the 1616 or 1616m system
There are five basic steps to installing your E-MU system:
1. Install the E-MU 02 CardBus card in your computer. Go there.
2. Install the PatchMix DSP software and drivers onto your computer.
3. Connect the MicroDock to the 02 CardBus Card using the supplied EDI cable.
4. Connect audio, MIDI and synchronization cables between the E-MU system and
your other gear.
Notes for Installation
• 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 2000 - SP 4, 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 “IKernel Application Error” on Windows XP: When installing this
software on Windows XP, you may be confronted with a “kernel error” 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.
Please read the following sections as they apply to your system as you install the E-MU
02, 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 02 CardBus 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.
E-MU 1616/1616M CardBus Digital Audio System
9
2 - Installation
Installing the CardBus Card and Software
Installing the CardBus Card and Software
Plug in the E-MU 02 CardBus Card
To plug the 02 CardBus Card into your computer
1. Turn on your computer and wait for it to finish loading Windows.
2. Insert the E-MU 02 CardBus card into the CardBus slot on your PC with the
symbol up. The “E” symbol should illuminate with a dim glow. The CardBus card
cannot be incorrectly inserted.
3. With CardBus card connected, continue to the software installation.
E Note: Once the E-MU
drivers are installed, the
E-MU symbol will glow
brightly whenever you
insert the CardBus card.
Software Installation
Installing the E-MU 02 Drivers
After installing the E-MU 02 CardBus card, you need to install the PatchMix DSP
software and E-MU 02 CardBus card drivers.
Windows 2000 or Windows XP
The software is not compatible with other versions of Windows.
1. As soon as you insert the CardBus card, Windows automatically detects it and
searches for device drivers.
2. When prompted for the audio drivers, click the Cancel button.
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 automatically. If not, from your Windows desktop, click Start->Run and type d:\setup.exe
(replace d:\ with the drive letter of your CD-ROM drive). You can also open the CD
and double-click 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
bottom of the 02 CardBus
Card.
4. The installation splash screen appears. Follow the instructions on the screen to
complete the installation.
5. Choose “Continue Anyway” when you encounter the “Windows Logo Testing”
warning screen. See the note on the following page for more information.
6. When prompted, restart your computer.
Uninstalling all Audio Drivers and Applications
At times you may need to uninstall or reinstall some or all of the 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 02 CardBus card device drivers or
applications.
10
Creative Professional
2 - Installation
Connecting the MicroDock
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.
+48V DC Adapter
Connecting the MicroDock
1. Connect the supplied EDI cable between the 02 CardBus Card and the MicroDock.
48 VDC
+
-
2. Connect the supplied +48 volt DC adapter to the+48VDC jack on the rear of the
Microdock. See the diagram at right.
3. Connect your audio inputs and outputs to the MicroDock as shown on page 18.
EDI
4. Turn the MicroDock on by turning the Headphone Volume control.
02 CardBus Card
The Headphone
Volume Control is
the Power Switch.
Connector Types
These connector types are used to connect the E-MU MIcroDockhardware components.
They will be referred to by the name shown in the first column of the following chart:
Name
Description
Connects
EDI
CAT5 Connector
02 CardBus 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)
Mic/Line Inputs
XLR Jacks or 1/4” jacks
(balanced or unbalanced)
XLR: connect to microphone
1/4”: instrument inputs or line inputs
Line In/Out
1/4” connectors
Connect to balanced or unbalanced
inputs and outputs.
Warning: The E-MU 02 CardBus card 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 02 CardBus card, or the MicroDock.
E-MU 1616/1616M CardBus Digital Audio System
11
2 - Installation
Connecting the MicroDock
12
Creative Professional
3 - CardBus Card & MicroDock
The E-MU 02 CardBus Card
3 - CardBus Card & MicroDock
The E-MU 02 CardBus Card
The E-MU 02 CardBus card is the heart of the system and contains E-MU’s powerful
E-DSP chip. The powerful hardware DSP on this little card leaves more CPU power free
on your computer for additional software plug-ins and other tasks.
CardBus Connector
Connect to Computer
D
ig
ita
lA
ud
io
Sy
st
em
E-MU 02 CardBus Card
EDI Connector
Connect to MicroDock
Monitor Output
Line Level or Headphones
Connections
CardBus Connector
Connects the E-MU 02 CardBus card to your computer.
Removing the CardBus Card
Before removing the CardBus card, you need to select “Safely Remove Hardware” from
the Ttaskbar. Otherwise ASIO channels will remain allocated to the Digital Audio
System and your other audio applications may develop problems or hang.
1. From the Taskbar, select the
icon. The “Safely Remove Hardware” pop-up
window appears.
2. Choose OK, then press the Eject button on the CardBus slot to eject the card.
EDI Connector
Connects to the MicroDock using the supplied EDI cable. This cable provides a a twoway data link between the E-MU 02 and the MicroDock.
Monitor Output
This output is designed to drive stereo headphones or any line-level input. Adjust the
monitor output level in the PatchMix DSP application to control the volume of this
output.
E-MU 1616/1616M CardBus Digital Audio System
13
3 - CardBus Card & MicroDock
The MicroDock
The MicroDock
The MicroDock connects to the E-MU 02 CardBus card via the EDI cable.
The MicroDock provides (4) balanced analog inputs, (2) microphone preamp inputs,
(6) balanced line-level analog outputs, (3) stereo 1/8” 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 “normalled” into line input 2L and
2R, 8 channels of ADAT digital input/output, and stereo S/PDIF digital input/output.
A Line
B Line
Mic
Clip
SL
Mic
Clip
-3
-6
-12
-20
SL
S/PDIF
48V
-3
-6
-12
-20
Out
In
Line Mic -
1L
1R
+50
+65
-15
0
2L
+50
+65
-15
0
2R
f The MicroDock is
completely “hot
pluggable”— It’s OK to
plug or unplug the
MicroDock while the
computer is turned on.
2L Phono 2R
Off
MIDI Cable
Gnd
48 VDC
+
-
In
It’s a good idea to
mute MicroDock inputs 2
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.
Out
1
2
3
EDI
Out
1L
1R
2L
2R
3L
3R
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)
(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:
14
(3)
stereo pairs of line level outputs
(1)
stereo pair driving a stereo headphone jack (Share the same routing as Line Outs 1L/1R)
(1)
stereo pair of S/PDIF/AES digital outputs
(4)
stereo pairs of ADAT channels on the ADAT optical output
(3)
stereo 1/8” 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
Creative Professional
3 - CardBus Card & MicroDock
The MicroDock
Front Panel Connections
1/4" Line
Level Input
Clip
Soft
Limit
-15dB to +50dB Gain
A Line
B Line
Mic
Clip
SL
S/PDIF
Phantom
Power
Coaxial I/O
On/Off
Signal
Meters
Mic
Clip
-3
-6
-12
-20
SL
S/PDIF
48V
-3
-6
-12
-20
1616
1616
In
Line Mic -
-15
0
Insert XLR Plug
for Mic Level
+50
+65
Headphone Volume
/ Power Switch
-15
0
Out
+50
+65
Input Gain
Controls
Off
ADAT
Optical I/O
Headphone
Output
0dB to +65dB Gain
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 microphones. 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.
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.
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 illuminated. 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.
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.
E-MU 1616/1616M CardBus Digital Audio System
15
3 - CardBus Card & MicroDock
The MicroDock
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 interleaving 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
88kHz or 96kHz
176kHz or 192kHz
8 channels of 24-bit audio
4 channels of 24-bit audio, using S/MUX standard interleaving
2 channels of 24-bit audio, using S/MUX standard interleaving
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.
The ADAT intputs and outputs can be configured in the System Settings (page 25) 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 due to the bandwidth limitations of
the optical components.
Note: PatchMix DSP does not support AC3 passthrough at this time.
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
thehigh-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.
16
Creative Professional
3 - CardBus Card & MicroDock
The MicroDock
Front Panel
Analog Connections
Mic
Use the 3-pin XLR jack
for Low Impedance
microphones.
A Line
B Line
Mic
Clip
SL
Mic
Clip
-3
-6
-12
-20
SL
S/PDIF
48V
-3
-6
-12
-20
Out
In
Line Mic -
+50
+65
-15
0
-15
0
+50
+65
On/Off
Off
& Phone Volume
Use the center
Phone Jack for
High Impedance
instruments such
as electric guitar
or bass.
Stereo
Headphones
Instrument
Digital Connections
Out
External A/D - D/A Converter
1
Optical
ADAT
(Optical)
In
2
3
4
5
6
7
8
(8 more analog inputs & outputs)
In
DAT or CD
Digital Audio Device with S/PDIF
Out
Out
S/PDIF
(Coax)
In
Out
Coaxial
In
Audio Outs
MIDI Out MIDI Keyboard
R E A L
T I M E
C O N T R O L L E R S
A S S I G N A B L E
K E Y S
P R E S E T
L E V E L
EXIT
ENTER
S A M P L E
P A G E
In
MIDI 1
Out
S E Q U E N C E R
P R E S E T S E L E C T
1
2
3
4
5
6
RETURN
7
8
9
0
.
EMULATOR
MIDI In
Audio Outs
MIDI Sound Module
In
MIDI 2
Out
E-MU 1616/1616M CardBus Digital Audio System
SAMPLE
I
TRANSPOSE
MASTER/GLOBAL
SAMPLE MANAGEMENT
MULTIMODE
PRESET MANAGEMENT
PRESET DEFINITION
TRIGGERS
DIGITAL PROCESSING
INC/YES
ABC
1
O
DEC/NO
PRESET
DYNAMIC PROCESING
ENTER
VOLUME
DRIVE SELECT
LOAD
SAVE
AUDITION
TRIGGER MODE
ESCAPE
GHI
DEF
2
3
JKL
MNO
4
5
6
PRS
TUV
WXY
7
8
9
QZ
0
MIDI
MIDI In
17
3 - CardBus Card & MicroDock
The MicroDock
Rear Panel Connections
4 Balanced Line Level Inputs
Turntable Inputs
(configured as 2 stereo pairs)
(tied to line input 2)
1L
1R
2L
2R
2L Phono 2R
Turntable
Ground
MIDI Port
Connector
48 Volt DC
Power Input
MIDI Cable
Gnd
48 VDC
+
-
In
Out
1
2
3
EDI
Out
1L
1R
2L
2R
3L
6 Balanced Line Level Outputs
(configured as 3 stereo pairs)
3R
Alternate Outputs
(same as outputs 1-3)
Connect to
E-MU 02 CardBus 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 85 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 3
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 1/4” adapters to connect line level signals to the line level analog
inputs.
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.
Line Level Analog Outputs
Six balanced 24-bit, line-level, analog outputs are provided (1-3). Output pair 1 is designated 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.
18
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.
Creative Professional
3 - CardBus Card & MicroDock
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
2 L/R
3 L/R
Tip = 1L Ring = 1R
Tip = 2L Ring = 2R
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 “Sounds and Audio Devices”.
There are two completely independent set 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 to the MicroDock to the E-MU 02 CardBus card using a CAT5-type computer
cable. The cable supplied with the MicroDock is specially shielded to prevent unwanted
RF emissions.
Basic
Connections
MIDI Synthesizer
Out
MIDI In
Turntable
MIDI 1
MIDI Out
In
Audio
from
Synthesizer
1L
1R
*
*
2L
2R
2L Phono 2R
MIDI Cable
Gnd
48 VDC
+
-
In
AC Adapter
Out
1
2
3
EDI
Out
1L
1R
2L
2R
3L
CardBus
Card
3R
Connect
Desktop
Speakers to
1/8" jacks
Audio
to
Monitors
Powered
Desktop
Speakers
Stereo
Mixer
&
Speakers
* Note: Line Inputs 2L/2R and Phono 2L/2R cannot be used at the same time.
E-MU 1616/1616M CardBus Digital Audio System
19
3 - CardBus Card & MicroDock
The MicroDock
20
Creative Professional
4 - The PatchMix DSP Mixer
PatchMix DSP
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 “Sessions” at will.
To Invoke the PatchMix DSP Mixer
on the Windows System Tray. The PatchMix
DSP mixer window appears.
1. Left-click once on the E-MU icon
Overview of the Mixer
Physical Input Strips
f Click on the buttons
and knobs in the mixer
screen below to jump to
the description of the
control.
Toolbar
ASIO Input Strip
Add New
Strip
Display
Select
Buttons
Delete
Strip
“TV”
Screen
Channel
Insert
Section
Pan
Controls
Aux
Effects
Section
Aux
Sends
Sync/
Sample
Rate
Indicators
Volume
Fader
Solo/Mute
Buttons
User
Definable
Scribble Strip
Monitor
Volume/Balance
/Mute Controls
WAVE Strip
Controls Windows Source Audio
(Direct Sound, Windows Media, etc.)
E-MU 1616/1616M CardBus Digital Audio System
Main
Inserts
Current
Session
Name
Main Mix
Output Volume
& Meters
21
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 “TV” 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
Input
Post-Fader Strip
Pre-Fader Strip
Insert
Section
Insert
Section
Mixer Block Diagram
Panning
Fader
Meter
MUTE
Aux 1
Aux 2
Aux
Bus 1
Aux
Bus 2
Aux 1
Send
Amount
Aux
Effects
Return
Amount
Insert
Section
Aux 2
Fader
MUTE
Main Bus
Monitor
Out
Return
Amount
Send
Amount
Insert
Section
MUTE
Main Bus
Effects
Insert
Section
Output 1L/1R
& Headphones
Monitor
Level
Main
Level
Main
Out
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.
22
Creative Professional
4 - The PatchMix DSP Mixer
E-MU Icon in the Windows Taskbar
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 PatchMix DSP application to start audio again.
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
Save
Session
Open
Session
“About”
PatchMix DSP
f Click the buttons in
the toolbar to learn about
their function.
Session
Settings
Show/Hide
Effects
Global
Prefs
New Session
Calls up the “New Session” dialog box. New Session.
Open Session
Calls up the standard “Open” dialog box, allowing you to open a
saved Session.
Save Session
Calls up the standard “Save” or “Save As…” dialog boxes, allowing
you to save the current Session.
Show/Hide Effects
Toggle button that shows or hides the FX palette.
Session Settings
Calls up the Sessions Settings window. Session Settings.
Global Preferences
Calls up the Global Preferences window.
About PatchMix DSP Right-Click on the E-MU logo to view the “About PatchMix DSP”
screen, which provides the software and firmware version
numbers and other information.
E-MU 1616/1616M CardBus Digital Audio System
23
4 - The PatchMix DSP Mixer
The Session
The Session
The current state of the PatchMix DSP mixer (fader settings, effects routings…everything!) 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.
Important: When
using any form of digital
input, you MUST
synchronize the Digital
Audio System to the
external digital device
(S/PDIF/ADAT).
PatchMix DSP comes with several session templates to choose from so when you create
a new session you can either create a “blank” 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.
New Session
You create a new session by clicking the “New Session” button in the PatchMix DSP
main Toolbar. The following dialog box appears.
Select a Template or new
Session at the desired
sample rate
Choose the location
where the new Session
will be saved.
Add your own
comment or note
about the Session
24
Creative Professional
4 - The PatchMix DSP Mixer
The Session
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
“Session Templates” folder (Program Files\Creative Professional\E-MU PatchMix
DSP\Session Templates). The system model number in parenthesis (1616) must precede
the template name in order to be recognized as a template.
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 contianing a different I/O configuration. Please see page 81 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.
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 “My Sessions” folder
is chosen by default.
Get in the habit of saving the session whenever you have created a special mixer setup.
This will make you 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.
E-MU 1616/1616M CardBus Digital Audio System
25
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.
E Note: if set to
“External” without an
external clock present,
PatchMix DSP defaults to
the internal 48kHz clock
rate.
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 “LOCKED” 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 “LOCKED” 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 “Cables - balanced or unbalanced?” in the Appendix for more
information.
Comparison of -10dBV & +4dBu Signal Levels
Consumer
Professional
(unbalanced)
(balanced)
}
+20 dBu
Clipping -->
Headroom
+ 6 dBV = +8 dBu
+ 2 dBV = +4 dBu
{
<-- Clipping
Headroom
-10 dBV = -8 dBu
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 “To Set the Input Levels of a Strip”
for details.
26
Creative Professional
4 - The PatchMix DSP Mixer
The Session
f Input too weak?
Use -10 Input setting.
Output too weak?
Use +4 Output setting
Input Level
Settings
Output Level
Settings
Optical
Input
Select
Optical
Output
Select
Mic Soft
Limiting
On/Off
S/PDIF
Output
Format
• 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
The Mic/Hi-Z inputs have built-in “soft limiters” which
automatically turn down the gain when the signal is about
to clip. The soft limiters allow you to record a hotter signal
without fear of clipping.
This control turns the soft limiters On or Off. See page 35
for additional information about the soft limiters.
• Optical Output Select
Selects between ADAT or optical S/PDIF for the MicroDock
ADAT Output. The coaxial S/PDIF Output is disabled when
S/PDIF optical is selected.
• S/PDIF Output Format
Selects between S/PDIF or AES/EBU format for S/PDIF. This
sets the S/PDIF-AES status bit, but does not affect the signal
level.
E-MU 1616/1616M CardBus Digital Audio System
27
4 - The PatchMix DSP Mixer
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
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:
f The Input Type will
turn RED if the input is not
available. (The MicroDock
may be disconnected.)
• Physical input = Hardware
(Analog/SPDIF/ADAT).
f Physical input strips
are shown with BLUE text.
• Host Input = Software
(Direct Sound, WAV, ASIO source)
f Host input strips are
shown with WHITE text.
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 Rightclicking.
Pan Controls
Aux Sends
Channel
Volume
Control
Mute/Solo
Buttons
Scribble Strip
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.
Volume 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.
This screen shows a mono strip on the left and a
stereo strip on the right.
28
Scribble Strips
Click inside the scribble strip and type
a name of up to eight characters.
Creative Professional
4 - The PatchMix DSP Mixer
Mixer Strip Creation
Mixer Strip Creation
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 hardware input or output such as a output jack.
f Adding or deleting a
strip “defragments” the
effect/DSP resources. If
you have used all your
effects and need another,
try deleting an unused
strip.
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:
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
Mixer Strip Type
Function
Physical: Dock Mic/Line
24-bit monophonic analog input from the MicroDock.
Physical: Dock In
24-bit stereo analog input from the MicroDock.
Physical: Dock S/PDIF
2 channel digital audio from the S/PDIF input on the MicroDock.
Physical: Dock ADAT
2 channel (x4 strips) digital audio from the ADAT input on the
MicroDock.
Host: WAVE L & R
Direct Sound, WDM, Windows Media (Sound generated or
handled by Windows, such as game sound, CD player, beep
sounds, etc.).
Host: From ASIO Out
2 channel digital audio from an ASIO source
(software application).
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.
f CDs & MP3s: The
WAVE strip is used to
playback CDs, Windows
Media Player, and nonASIO audio applications.
f See “Pre or Post Fader
Aux Sends” on page 39.
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. See Overview of the Mixer
E-MU 1616/1616M CardBus Digital Audio System
29
4 - The PatchMix DSP Mixer
Mixer Strip Creation
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 “The Effects Palette”. 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
Sends the signal to a selected ASIO host input, then returns a selected
ASIO host output to the chain.
Ext. Send/Return
Sends signal to a selected external output, then returns it to the chain
via a physical input.
External Send
Sends the signal to an external output. See “To Add a Send Insert:”.
Peak Meter
Peak meters allow you to monitor the signal level anywhere in the
chain. See “Meter Inserts”.
Trim Pot
You can insert a gain control with up to 30 dB of gain or attenuation.
A peak level meter and phase inverter are also included.
See “Trim Pot Insert”.
Test Tone
This special insert outputs a calibrated sine wave or noise source,
which can be used to track down audio problems.
See “Test Tone/Signal Generator Insert”.
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.
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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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.
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 “ASIO Send” becomes an input on your recording application, a
“Physical Out” 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
To ASIO, WAV or
Physical Output
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 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 “Insert Send/Return (Physical Output and Input)” from the list of options.
The following dialog box appears.
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Input
To Physical Output
From Physical Input
Insert
Send/Return
Panning
Fader
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.
Aux 1 Bus
Aux 2 Bus
Main Output Bus
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.
Input
Direct Mon
Recording
Input
Recording
Software
Direct Mon
Recording
Software
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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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 “peak hold” 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. A numeric readout above the meter shows the peak-hold level in dB.
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
70
60
50
40
30
20
10
--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.
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.
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 “Effect” button at the top of the TV screen.
The “I/O Settings” 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 “-10” if the input signal is too weak
and “+4” if the signal is too strong.
Digital audio has NO headroom past 0dBFS (FS = Full Scale) and will “hard clip” if the
signal exceeds 0dB. 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 “Trim Pot” 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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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 instruments.
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
Signal Type
(Sine wave, White or Pink Noise)
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.
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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Managing Your Inserts
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.
f Tip: Select the Insert
and press the Delete key
to delete the plug-in from
the strip.
3. The insert(s) are deleted from the insert chain.
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.
To Bypass All Inserts:
All Inserts in a strip can be 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.
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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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.
Input
Sidechain Diagram
(Post-Fader Aux Sends)
Pan
Fader
Mute
Send
Amount
Amt
Aux Bus 1
Return
Amount
Side
Chain
Send
Amount
Amt
Aux Bus 2
Return
Amount
Side
Chain
Main / Monitor Bus
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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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
Pre-Fader Aux Send
Volume Fader & Mute does NOT affect Send Levels
Pan
Send
Amount
Aux Bus 1
Send
Amount
Return
Amount
Side
Chain
Aux Bus 2
Fader
Return
Amount
Side
Chain
Amt
Amt
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.
Mute
Main / Monitor Bus
Input
Output
Post-Fader Aux Send
Volume Fader & Mute affects both Aux Send Levels
Pan
Fader
Mute
Send
Amount
Amt
Aux Bus 1
Return
Amount
Side
Chain
Send
Amount
Amt
Aux Bus 2
Return
Amount
Side
Chain
Main / Monitor Bus
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Level, Pan, Solo & Mute Controls
Pan Controls
Aux Send
Amount
Controls
The Pan control comes before the Level Control
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 mixer’s
Level Control
output. 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
Mute & Solo
channel reverts to being muted.
Buttons
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.
Scribble Strip
40
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. “vocals”, “bass”,
“drums” and so on.
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4 - The PatchMix DSP Mixer
Main Section
Main Section
Physical/Host
Select Buttons
View
Selection
Buttons
“TV” Screen
Aux
Insert
Section
Master
Aux Send
Amounts
Main
Insert
Section
Master Aux
Return
Amounts
Sync &
Sample Rate
Indicators
Monitor Controls
Output
Fader &
Meters
Session Name
The main section contains all controls for controlling the main mix elements as well as
a “TV screen” 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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Main Section
TV Screen & Selectors
The “TV screen” 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 “No
Insert”.
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 “List of Core Effects” for detailed information about the individual effects.
Effect Display
View Button
E Note: Effects have to
be placed into an insert
location before you can
program them.
Effect Location
Effect Bypass &
Solo Buttons
Wet/Dry Mix Control
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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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
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 “physical” 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 “Insert Section” for information on how to connect the inserts.
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Main Section
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.
Send
Amount
Return
Amount
Input
Input
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.
Sidechain Diagram
(Post-Fader Aux Sends)
Pan
Fader
Mute
Aux
Amt
Sidechain
Effects
Aux
Amt
Send
Amount
Aux Bus
Return
Amount
Side
Chain
Main Bus
Output
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 “Insert Section” for more information.‚
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 “EXTERNAL”.
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
“LOCKED” 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 or the higher rates, the hardware automatically switches to internal 48kHz clock until the proper external clock is restored. The
“LOCKED” LED will be off and the two units are NOT synchronized. Always check the
“LOCKED” LED when using an external clock source to make sure you are samplelocked.
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Main Section
Output Section
Clip Indicators
Main Output Level Fader
Sync/Sample
Rate Indicators
Main
Insert
Section
Monitor
Mute
Monitor
Balance
Output Level
Meters
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.
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.
MAIN MIX
0dB
10
10
20
20
30
30
40
40
50
50
L
R
-12dB
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.
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.
E-MU 1616/1616M CardBus Digital Audio System
45
4 - The PatchMix DSP Mixer
Main Section
46
Creative Professional
5 - Effects
Overview
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 “grayed out” 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.
f Saving a session
“defragments” the effect/
DSP resources. If you
have used all your effects
and need another, try
saving the session.
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 “Core Effects” folder contains the effect algorithms
themselves. This folder cannot be modified. The other folders contain “Effects Chains”,
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 “FX Insert Chains” on page 48.
New Folder Icon
Effect Categories
Core Effects
Multi-Effects
Distortion Lo-fi
Drums & Percussion
Environment
Equalization
Guitar
Morpher
Multi Effects
Reverb
Synths & Keys
Vocal
E-MU 1616/1616M CardBus Digital Audio System
47
5 - Effects
The Effects Palette
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.
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.
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.
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 “Delete Insert(s)” from the top of the list. The effect will be deleted.
FX Insert Chains
FX Insert Chains can be used to save several effects and their settings into a single multieffect. 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.
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 “Save FX Insert Chain”. The New FX preset dialog box appears.
f Trim pots, peak meters
and test tone generators
will also be included in
the FX chain.
5. Select a category folder where your preset will be placed, and enter a new preset
name for your 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.
48
Creative Professional
5 - Effects
The Effects Palette
Creating, Renaming & Deleting Categories or Presets
There are several utilities to help you organize your effects presets.
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 “Enter the Name of the New Category.”
• Alternatively, you can Right-click over an Effects Folder, which calls a pop-up
dialog box with the option to “Create New Category.”
2. Type in a name for your new folder.
3. Click OK to create a new folder or Cancel to cancel the operation.
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 “Delete Category”. 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.
To Rename an Effects Category
1. Right-click on the category folder you wish to rename. A pop-up selection box
appears.
2. Select “Rename Category”. A pop-up dialog box appears, asking you to “Enter New
Category Name.”
3. Click OK to rename the folder or Cancel to cancel the operation.
E-MU Digital Audio System
49
5 - Effects
FX Edit Screen
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 “No Insert”.
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.
E Note: Effects have to
be placed into an insert
location before you can
program them.
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.
Effects Display
View Button
Effect Location
Effect Bypass &
Solo Buttons
Wet/Dry Mix Control
Effect Parameters
User Preset Section
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)
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 “Bypass Insert” from the list of options. The insert effect name will “gray-out”
to indicate that the insert effect is bypassed.
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.
50
Creative Professional
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 “Solo Insert” from the list of options. The other Insert Effect names in the
strip will “gray-out” to indicate that they are bypassed.
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 “Bypass All Inserts” from the list of options. All the insert names will be
“grayed-out” to indicate that they are bypassed.
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 “Un-Bypass All Inserts” 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.
E To copy or share User
Presets, you must save
them as FX Palette
effects.
Click here for Edit Menu
Click here to Select Presets
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.
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-MU Digital Audio System
51
5 - Effects
FX Edit Screen
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.
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.
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.
Core Effects and Effects Presets
The Core Effects cannot be removed or copied. Effect presets (stored in “C:\Program
Files\Creative Professional\E-MU 1616\E-MU PatchMix DSP\Effect Presets”) can be
copied, e-mailed or shared like any other computer file.
52
f Hint: You can open
the effects presets with
“NotePad” or other word
processor to view and
edit the name and
parameters.
Creative Professional
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.
f Tip: Saving a session
“defragments” the effect/
DSP resources. If you
have used all your effects
and need another, try
saving the session.
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.
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.
Stereo Reverb
2
4-Band EQ
No.
Example 3
No.
Lite Reverb
5
Stereo Reverb
1
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
21
Total Effects
22
E-MU Digital Audio System
Example 2
Total Effects
18
53
5 - Effects
Core Effects Descriptions
Core Effects Descriptions
1-Band Para EQ
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.
+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
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
Corner
Freq
Boost
+
Cut
+15dB
Gain
or…
-
-15dB
High Shelf
Corner
Freq
Frequency
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.
Range: -15dB to +15dB
Corner Frequency Sets the frequency where the signal begins getting cut or boosted
with the Gain control. Range: 80Hz to 16kHz
54
Creative Professional
5 - Effects
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
Corner
Freq.
Boost
+
Cut
+24dB
Gain
Mid Band
-
High Shelf
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
-24dB
Frequency
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 “zero-in” 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
E-MU Digital Audio System
55
5 - Effects
Core Effects Descriptions
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 55.
Low-Shelf
Mid 1-Band
Mid 2-Band
Corner
Frequency
Boost
Cut
Gain
Corner
Frequency
+
-
High-Shelf
Width
Width
Center
Frequency
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
56
Creative Professional
5 - Effects
Core Effects Descriptions
Auto-Wah
This effect creates the sound of a guitar wah-wah pedal. The “Wah” 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 “envelope”, 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
Sweep Range
Input
Wave
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 “wah” sweep.
Range: 0ms to 500ms
Env. Release Time
Sets the ending or release rate of the “wah” sweep.
Range: 10ms to 1000ms
Sweep Range
Controls the amount of “wah” 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
E-MU Digital Audio System
57
5 - Effects
Core Effects Descriptions
Chorus
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°
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 “hotter” 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.
In
Signal path = Stereo
Delay
VCA
Out
Level
Control
Threshold
Ratio
58
Post Gain
Release
Attack
Creative Professional
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 “pumping” 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 “look into the future” up to 4 milliseconds 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 “how
much” compression will be applied. Range: 1:1 to ∞:1
Post Gain
Amplifies the signal after it has been compressed to bring up the
volume. Range -60dB to +60dB
E-MU Digital Audio System
59
5 - Effects
Core Effects Descriptions
Parameter
Description
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 “fuzz box” 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 “how
much” 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!
Lowpass
Filter
Bandpass
Filter
Distortion
In
Out
Signal path = Stereo
LP Filter
Cutoff
Edge
EQ BW
Gain
EQ Center
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
60
Creative Professional
5 - Effects
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 cancellations 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
Out
Signal path = Stereo
Delay
LFO
Phase
Waveform
Parameter
Description
Delay
Sets the initial delay of the flanger in .01 millisecond increments.
This parameter allows you to “tune” 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°
E-MU Digital Audio System
61
5 - Effects
Core Effects Descriptions
Freq Shifter
This unusual effect is sometimes called “spectrum shifting” or “single sideband
modulation”. 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 “musical” 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.
f You can also type in
exact frequencies to a
resolution of 1/10 Hz.
Comparison between Pitch and Frequency Shifting
Harmonic
Original Pitch Shifted Frequency Shifted
(Hz)
(100 Hz)
(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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Creative Professional
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 “lookahead delay” 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
E-MU Digital Audio System
Amplifies the signal after it has been compressed to
bring up the volume. Range 0dB to 36dB
63
5 - Effects
Core Effects Descriptions
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.
Early Reflections
Reverberation
Time
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 “boominess” 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
frequencies last longer as the percentage is increased.
Range: 0% to 100%
LF Decay Factor
Sets the rate at which low frequencies die away. The low
frequencies last longer as the percentage is increased.
Range: 0% to 100%
Early Reflections
Sets the volume of the initial wall reflections.
Range: 0% to 100%
Reverberance
Sets the amount of scattering of the early reflections and
the reverberation cloud. Range: 0% to 100%
64
Creative Professional
5 - Effects
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 filtering: See page 61.
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.
Feedback
HF
Rolloff
L Out
L In
Delay
R In
R Out
Delay Time
Parameter
Description
Delay Time
Sets the length of the delay in milliseconds.
(.01ms. minimum increment between settings)
Mono Delay 100 Range: 1 millisecond to 100 milliseconds
Mono Delay 250 Range: 1 millisecond to 250 milliseconds
Mono Delay 500 Range: 1 millisecond to 500 milliseconds
Mono Delay 750 Range: 1 millisecond to 750 milliseconds
Mono Delay 1500 Range: 1 millisecond to 1.5 seconds
Mono Delay 3000 Range: 1 millisecond to 3 seconds
Feedback
Sets the amount of delayed signal that will be recirculated through
the delay line. Range: 0% to 100%
High Freq. Rolloff
Damps high frequencies in the feedback path.
Range: 0% to 100%
E-MU Digital Audio System
65
5 - Effects
Core Effects Descriptions
Phase Shifter
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
Shifter
Signal path = Stereo
LFO Center
Out
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.
66
Creative Professional
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.
E-MU Digital Audio System
67
5 - Effects
Core Effects Descriptions
Stereo Delays - 100, 250, 500, 750, 1500
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 roboticsounding effects. Using the longer stereo delays, you can “overdub” musical lines one
on top of the other with the feedback control turned up.
Feedback
HF
Rolloff
In
Delay
Out
Signal path = Stereo
L Delay R Delay
Time
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
(.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
Feedback
Sets the amount of delayed signal that will be recirculated through
the delay line. Range: 0% to 100%
High Freq. Rolloff
Damps high frequencies in the feedback path. Range: 0% to 100%
68
Creative Professional
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.
Early Reflections
Late Reverb
Time
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 “boominess” of the room.
Parameter
Description
Decay Time
Sets the length of the Late Reverb. Range 1.5 to 30seconds
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
E-MU Digital Audio System
69
5 - Effects
Core Effects Descriptions
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
Frequ
Tim
e
ency
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
70
Creative Professional
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
FX Palette
FX Inserts
Input Signal Present
Output Signal Present
FX Parameters
FX Presets
Preferences
f Cubase SX/SL/LE 2.0,
Nuendo and Sonar (using
the Cakewalk VST
adapter 4.4.1) implement
VST 2.X auto delay
compensation.
Preset Editing
Parameter
Description
PowerFX On/Off
Enables or bypasses E-MU PowerFX.
FX Palette
Select from a single “Core” 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 “User Preset Section” for more information
E-MU Digital Audio System
E-MU PowerFX are not
available at 88.2kHz,
96kHz, 176.4kHz and
192kHz sample rates.
71
5 - Effects
E-MU PowerFX
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, SoundForge).
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.
Using any driver other
than “E-MU ASIO” may
produce undesirable
results when using E-MU
PowerFX.
Setup E-MU PowerFX
4. Make sure the blue button
is illuminated, indicating that E-MU Power FX is on.
The blue “Signal Present” 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 “User Preset Section” for more information.
Add Delay Compensation (if needed)
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.
72
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.
Creative Professional
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 “Write Automation” is activated in Cubase, control changes
made in the PowerFX window during playback will be recorded on a special
“Automation Subtrack”. When “Automation Read” is activated, the recorded control
changes will be played back.
Steinberg Cubasis
does not have the
control automation
feature.
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 “Automation Write” 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 “On”
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 “Automation Write” and enable “Automation Read”
. Playback the song
to hear and view your changes.
7. To edit Automation, first enable both “Automation Write” and “Automation Read”
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.
Once you have
recorded or drawn
automation, do not
delete or move effects
from the Insert Strip.
Doing so will result in
unpredictable behavior.
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.
E-MU PowerFX Resource Availability
Because different collections of VST plug-ins and PatchMix Sessions can be run simultaneously, 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 “redded out”.
• 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 “grayed out”.
• 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.
E-MU Digital Audio System
73
5 - Effects
E-MU PowerFX
E-MU PowerFX Compatibility Chart
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
Off
Off
Steinberg Cubase LE
Yes
Off
Off
Steinberg Cubase SL
Yes
Off
Off
Steinberg WaveLab 4
Yes
On
Off
Steinberg WaveLab Lite (ver 4)
Yes
On
Off
Steinberg WaveLab 5
No
On
Either
Sony Acid 4
Yes
On
Off
Sony Vegas 5
Yes
On
Off
Sony SoundForge 7
No
Power FX
crashes when
launched.
On
Off
Adobe Audition 1.5
No
Audio
distortion &
immediate
lockup.
Any
Any
FruityLoops Studio 4.5
Yes
Off
On
Abelton Live 3.5
No
On
Off
Cakewalk Sonar 3
Yes
Off
Off
74
Instrument
Freeze triggers
error if
not in render
mode.
Pops & clicks
may occur.
(Try 8 buffers at
1024)
Distortion
when FX
parameters are
changed.
Creative Professional
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 “ASIO Buffer Latency” 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 “Realtime Render” 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 “Freezing” 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 “Render Mode” box in the E-MU PowerFX preferences. See page 72.
• We recommend that you only use the MME/WAVE E-DSP Wave [xxxx] drivers.
• Reduce the “Buffer Size” 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.
E-MU Digital Audio System
75
5 - Effects
E-MU VST E-Wire
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 compensates 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.
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 delaycompensation 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 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.
E-Wire VST plug-in
Send to Strip
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.
76
Creative Professional
5 - Effects
E-MU VST E-Wire
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
“New Session”, select “E-Wire Example” 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 compensated 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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5 - Effects
E-MU VST E-Wire
Currently automatic delay compensation is supported by the Steinberg 2.0 family
(Nuendo 2.x, Cubase SX 2.0, Cubase LE 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 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 “dry” 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 Compensator 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
Track 1
Track 2
Track 3
Insert
Insert
Insert
E-Wire
E-Delay
E-Delay
E-Delay Compensator Use
PatchMix
DSP
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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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 “just in case” 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
Track 2
Track 3
Insert
Insert
Insert
Insert
Insert
PowerFX
or E-Wire
PowerFX
or E-Wire
PowerFX
or E-Wire
E-Delay
1
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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E-MU VST E-Wire
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Creative Professional
6 - Using High Sample Rates
Overview
6 - 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. These changes are summarized in the following
tables. 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:
• Effect processors are disabled. (Output sends & returns are still available.)
• ADAT is reduced to 4 channels at 88k/96k, and 2 channels at
176k/192kHz.
• ASIO channels are reduced to 8 stereo ASIO channels at 88k/96k,
and 4 stereo ASIO 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 Sonorus® S/MUX™ 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.
E-MU 1616 System at 176kHz or 192kHz
At the 176.4kHz or 192kHz sample rates you have 4 inputs and 10 output channels.
There are three possible input configurations when using the E-MU 1616 system at these
high sample rates.
1. Keep all Analog I/O, but lose S/PDIF
2. Keep all Analog I/O, but lose ADAT
3. Keep S/PDIF & ADAT, but lose Line Inputs 2L/2R & Line Outputs 3L/3R
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.
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 - Using High Sample Rates
Overview
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
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Overview
E-MU 1616 Inputs & Outputs at 176.4k or 192k
Source
Input
Analog
&
S/PDIF
Output
Analog &
S/PDIF
Input
S/PDIF
& ADAT
Output
S/PDIF
& ADAT
Input
Analog
& ADAT
Output
Analog
& ADAT
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)
-------
2
-------
0
-------
2
S/PDIF
2
2
2
2
0
0
Total
8
8
8
8
8
8
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 “bit accurate” 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-nativesample-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.
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Overview
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7 - Appendix
Useful Information
7 - Appendix
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
2
1
1
2
3
3
Input
Output
1 = Ground/shield
2 = Hot (+)
3 = Cold (-)
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.
Sleeve = Ground
Balanced 1/4”
TRS Connectors
Tip = Hot (+)
Ring = Cold (-)
Unbalanced 1/4”
Connectors
Sleeve = Ground
Tip = Signal
Balanced cables have one ground (shield) connection and two signal-carrying
conductors of equal potential but opposite polarity. There is one “hot” or positive lead,
and a “cold” 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 “hot” 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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7 - Appendix
Useful Information
Adapter Cables
1/8” Mini-phone to 1/4” Adapters
To connect headphones with an 1/8” (mini-phone) plug to the headphone jack on the
MicroDock, you need a 1/8” to 1/4” adapter. These handy devices are available at
electronic department stores everywhere.
1/8" to 1/4"
Headphone Adapter
Headphones with
1/8" plug
Cinch (RCA) to 1/4” 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.
Cinch/RCA
Plug
Tip = Hot (+)
1/4" Phone Plug
Shaft = Ground
Tip = Hot (+)
Sleeve = Ground
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 02 CardBus card. This cable may also work to connect S/PDIF out
from the 02 CardBus card to the AES/EBU input of other digital equipment.
From AES/EBU
Device
1
2
N.C.
+
To S/PDIF
In
3
-
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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 “rides” 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.
1
2
(grd)
3
Balanced dynamic microphones are not affected by phantom
power. An unbalanced dynamic microphone may not work
properly, but will probably not be damaged if phantom power
is left on.
+48V
Ribbon microphones should NOT be used with phantom
power on. Doing so can seriously damage the ribbon element.
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.
Appearance Settings in Windows
Adjusting the “Performance Options” in Windows will improve the screen appearance
when moving the mixer around on the screen.
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
Technical Specifications
Specifications: 1616m System
GENERAL
Sample Rates
44.1 kHz. 48 kHz, 96 kHz, 192 kHz from internal crystal
Accepts externally supplied clock from S/PDIF or ADAT
Bit Depth
16 or 24-bits
Hardware DSP
100MIPs custom audio DSP.
Zero-latency direct hardware monitoring with effects
Converters & OpAmps
ADC - AK5394 (AKM)
DAC - CS4398 (Cirrus Logic)
OpAmp - NJM2068M, NJM2122, NJM2082 (JRC)
WDM Drivers
Stereo — operational at 44.1kHz, 48kHz, 88.2kHz, 96kHz,
176.4kHz & 192kHz
MicroDockm Power Use
.32A @ +48VDC
15.4 Watts
ANALOG LINE INPUTS
Type
Servo-balanced, DC-coupled, low-noise input circuitry
Level (software selectable)
Professional: 18 dBV maximum (balanced)
Consumer: 6 dBV maximum (unbalanced)
Frequency Response
20 Hz - 20 kHz, +0.0/-0.03 db
THD + N
-110 dB (.0003%) (-1 dBFS, 20kHz BW)
SNR
120 dB (A-weighted, 20kHz BW)
Dynamic Range
120 dB (A-weighted, 20kHz BW)
Channel Crosstalk
-120 dB, (1 kHz)
Common-mode Rejection
-79 dB at 60Hz
Input Impedance
10K ohm
ANALOG LINE OUTPUTS
Type
Balanced, low-noise, 2-pole low-pass differential filter,
AC-coupled
Level (software selectable)
Professional: 18 dBV maximum (balanced)
Consumer: 6 dBV maximum (unbalanced)
Frequency Response
20 Hz - 20 kHz, +0.0/-0.03 dB
THD + N
-105 dB (.0005%) (-1dBFS, 20kHz BW)
SNR
120 dB (A-weighted, 20kHz BW)
Dynamic Range
120 dB (A-weighted, 20kHz BW)
Stereo Crosstalk
< -115 dB, 1kHz
Output Impedance
560 ohms
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Technical Specifications
Specifications: 1616m System
MIC PREAMP/LINE INPUT
HI-Z LINE INPUT
Gain Range:
-15 to +51 dB
Max Level:
19 dbV (21.2 dBu)
THD+N:
-105 dB (.00056%), -1 dBFS, 20kHz BW
SNR:
118 dB, (A-weighted, 20kHz BW)
Dynamic Range:
118 dB, (A-weighted, 20kHz BW)
Crosstalk:
Minimum Gain: -140 dB (1kHz)
Maximum Gain: -80 dB (1kHz)
Frequency Response:
+0.10/-0.00 dB, 20 Hz - 20 kHz
Input Impedance:
1M ohm
CMRR:
-44 dB (60Hz)
MICROPHONE PREAMP
Gain Range:
-1.3 to +65 dB
Max Level:
-5.6 dbV (7.8 dBu)
THD+N:
Min Gain: -112dB (.00025%), (1kHz, -1 dBFS, 20kHz BW)
40dB Gain: -96dB (.0015%)
60dB Gain: -75dB (.017%)
SNR:
119 dB (A-weighted, min. gain)
Frequency Response:
20 Hz - 20 kHz ±0.08 dB (20Hz - 20kHz, gain +40dB)
Input Impedance:
1.5 Kohm
CMRR:
-95 dB (60Hz, 35 dB gain, -1 dBFS)
Crosstalk:
-100 dB (1 kHz, 40 dB gain, -1 dBFS)
HEADPHONES
Frequency Response:
+0.0/-0.07 dB, 20 Hz - 20 kHz
THD+N: (1 kHz, max. level)
24 ohm load: -85 dB (0.018%)
600 ohm load: -94 dB (0.002%)
SNR:
118 dB (A-weighted)
Dynamic Range:
118.5 dB (A-weighted)
Stereo Crosstalk:
24 ohm load: < -43 dB (1kHz at 0 dBFS)
600 ohm load: < -100 dB (1kHz at 0 dBFS)
Max Output Power:
100 mW (24Ω load)
Output Impedance:
22 ohms
Gain Range:
85 dB
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Technical Specifications
Specifications: 1616m System
CARD HEADPHONE AMP
Frequency Response:
+0.05/-0.0 dB, 20 Hz - 20 kHz
THD+N:
24 ohm load: -80 dB (0.01%)
65 ohm load: -85 dB (0.0056%)
600 ohm load: -96 dB (0.0016%)
SNR:
116 dB (A-weighted, 22kHz BW)
Dynamic Range:
116 dB (A-weighted, 22kHz BW)
Stereo Crosstalk:
-99 dB (1kHz at -1 dBFS, 660 ohm load)
Max Output Power:
50 mW (24Ω load)
Output Impedance:
22 ohms
TURNTABLE INPUT
RIAA equalized phono input
Frequency Response:
+/-0.5 dB, 50 Hz - 20 kHz
Deviation from RIAA
+0.2/-0.3 dB (50Hz to 15kHz)
THD+N:
-92 dB (.0025%) (20kHz BW, unbalanced)
SNR:
107 dB (A-wt, 20kHz BW)
Stereo Crosstalk:
< -80 dB (1kHz at -1 dBFS)
Maximum Level:
Professional: 60 mV RMS
Consumer: 15 mV RMS
Input Capacitance:
220 pF
Input Impedance:
47K ohm
DIGITAL I/O
S/PDIF
• 2 in/2 out coaxial (transformer coupled)
• 2 in/2 out optical (software switchable with ADAT)
• AES/EBU or S/PDIF (switchable under software control)
ADAT
• 8 channels, 24-bit @ 44.1/48 kHz
• 4 channels, 24-bit @ 96 kHz
• 2 channels, 24-bit @ 192 kHz
MIDI
2 MIDI in, 2 MIDI out
SYNCHRONIZATION
Internal Crystal Sync:
44.1kHz, 48 kHz, 88.2kHz, 96 kHz, 176.4kHz, 192 kHz
ADAT, S/PDIF (optical or coaxial)
SRSync SourceRMS jitter in picoseconds
RMS JITTER @ 44.1K
(Measured via Audio Precision 2) 44.1 kHz internal Crystal 596ps
44.1 kHz Optical Input
795ps
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7 - Appendix
Technical Specifications
Specifications: 1616 System
GENERAL
Sample Rates
44.1 kHz. 48 kHz, 88.2kHz, 96 kHz, 176.4kHz & 192 kHz
from internal crystal.
Accepts externally supplied clock from S/PDIF or ADAT
Bit Depth
16 or 24-bits
Hardware DSP
100MIPs custom audio DSP.
Zero-latency direct hardware monitoring with effects
Converters & OpAmps
ADC - PCM1804 (TI)
DAC - CS4392 (Cirrus Logic)
OpAmp - NJM2068M, NJM2122, NJM2082 (JRC)
WDM Drivers
Stereo — operational at 44.1kHz, 48kHz, 88.2kHz, 96kHz,
176.4kHz & 192kHz
MicroDock Power Use
.20A @ +48VDC
10 Watts
ANALOG LINE INPUTS
Type
Servo-balanced, DC-coupled, low-noise input circuitry
Level (software selectable)
Professional: 18 dBV maximum (balanced)
Consumer: 6 dBV maximum (unbalanced)
Frequency Response
20 Hz - 20 kHz, +0.05/-0.05 db
THD + N
-102 dB (.0007%) -1 dBFS, 20kHz BW
SNR
110 dB (A-weighted, 20kHz BW)
Dynamic Range
110 dB (A-weighted, 20kHz SPCL)
Channel Crosstalk
< -120 dB, (1 kHz)
Common-mode Rejection
-50 dB at 60Hz
Input Impedance
10K ohm
ANALOG LINE OUTPUTS
Type
Balanced, low-noise, 2-pole low-pass differential filter,
AC-coupled
Level (software selectable)
Professional: 18 dBV maximum (balanced)
Consumer: 6 dBV maximum (unbalanced)
Frequency Response
20 Hz - 20 kHz, +0.0/-0.08 dB
THD + N
-98 dB (.0012%) (-1dBFS, 20kHz BW)
SNR
112 dB (A-weighted, 20kHz BW SPCL)
Dynamic Range
112 dB (A-weighted, 20kHz SPCL)
Stereo Crosstalk
< -115 dB, 1kHz
Output Impedance
560 ohms
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Technical Specifications
Specifications: 1616 System
MIC PREAMP/LINE INPUT
HI-Z LINE INPUT
Gain Range:
-15 dB to +51 dB
Max Level:
19.5 dbV
THD+N:
-101 dB (.0009%), -1 dBFS, 20kHz BW
SNR:
110 dB, (A-weighted, 20kHz BW)
Dynamic Range:
110 dB, (A-weighted, 20kHz BW)
Crosstalk:
Minimum Gain: -140 dB (1kHz)
Maximum Gain: -80 dB (1kHz)
Frequency Response:
+0.0/-0.06 dB, 20 Hz - 20 kHz
Input Impedance:
1M ohm
CMRR:
-75 dB (60Hz)
MICROPHONE PREAMP
Gain Range:
-2 dB to +64 dB
Max Level:
-6.5 dbV
THD+N:
Min Gain: -102 dB (.0008%), (1kHz, -1 dBFS, 20kHz BW)
40dB Gain: -91dB (.0029%)
60dB Gain: -69 dB (.035%)
SNR:
110 dB (A-weighted, min. gain)
Frequency Response:
20 Hz - 20 kHz ±0.08 dB (20Hz - 20kHz, gain +40dB)
Input Impedance:
1.5 Kohm
CMRR:
-67 dB (60Hz, 35 dB gain, -1 dBFS)
Crosstalk:
-100 dB (1 kHz, 40 dB gain, -1 dBFS)
HEADPHONE AMP
Frequency Response:
+0.0/-0.07 dB, 20 Hz - 20 kHz
THD+N: (1 kHz, max. level)
24 ohm load: -80 dB (0.01%)
600 ohm load: -94 dB (0.002%)
SNR:
112 dB (A-weighted, 20kHz BW)
Dynamic Range:
112 dB (A-weighted, 20kHz BW)
Stereo Crosstalk:
24 ohm load: < -43 dB (1kHz at 0 dBFS)
600 ohm load: < -90 dB (1kHz at 0 dBFS)
Max Output Power:
100 mW (24Ω load)
Output Impedance:
22 ohms
Gain Range:
85 dB
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Technical Specifications
Specifications: 1616 System
CARD HEADPHONE AMP
Frequency Response:
+0.05/-0.0 dB, 20 Hz - 20 kHz
THD+N:
24 ohm load: -80 dB (0.01%)
65 ohm load: -85 dB (0.0056%)
600 ohm load: -96 dB (0.0016%)
SNR:
116 dB (A-weighted, 22kHz BW)
Dynamic Range:
116 dB (A-weighted, 22kHz BW)
Stereo Crosstalk:
-99 dB (1kHz at -1 dBFS, 660 ohm load)
Max Output Power:
50 mW (24Ω load)
Output Impedance:
22 ohms
TURNTABLE INPUT
RIAA equalized phono input
Frequency Response:
+/-0.5 dB, 50 Hz - 20 kHz
Deviation from RIAA
+0.2/-0.3 dB (50Hz to 15kHz)
THD+N:
-90 dB (.003%) (20kHz BW, unbalanced)
SNR:
96 dB (A-wt, 20kHz BW)
Stereo Crosstalk:
< -80 dB (1kHz at -1 dBFS)
Maximum Level:
Professional: 60 mV RMS
Consumer: 15 mV RMS
DIGITAL I/O
S/PDIF
• 2 in/2 out coaxial (transformer coupled)
• 2 in/2 out optical (software switchable with ADAT)
• AES/EBU or S/PDIF (switchable under software control)
ADAT
• 8 channels, 24-bit @ 44.1/48 kHz
• 4 channels, 24-bit @ 96 kHz
• 2 channels, 24-bit @ 192 kHz
MIDI
2 MIDI in, 2 MIDI out
SYNCHRONIZATION
Internal Crystal Sync:
44.1kHz, 48 kHz, 88.2kHz, 96 kHz, 176.4kHz, 192 kHz
ADAT, S/PDIF (optical or coaxial)
SRSync SourceRMS jitter in picoseconds
RMS JITTER @ 44.1K
(Measured via Audio Precision 2) 44.1 kHz internal Crystal 596ps
44.1 kHz Optical Input
795ps
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Technical Specifications
Dimensions & Weight
MICRODOCK
MicroDock Weight:
2.27 lb / 1.03 kg
Dimensions:
W: 7.25" H: 1.625" L: 7.75"
W: 184 mm H: 41 mm L: 196 mm
02 CardBus Card
Weight:
0.095 lb / 0.043 kg
Dimensions:
W: 2.125" H: .5625" L: 4.675"
W: 54 mm H: 14 mm L: 117 mm
94
Creative Professional
7 - Appendix
Internet References
Internet References
The internet contains vast resources for the computer musician. A few useful sites are
listed here, but there are plenty more. Check it out.
Software Updates, Tips & Tutorials................ http://www.emu.com
Setting up a PC for Digital Audio .................. http://www.musicxp.net
MIDI Basics....................................................... Search for “MIDI Basics” (many sites)
MIDI & Audio Recording ................................ http://www.midiworld.com
MIDI & Audio Recording ................................ http://www.synthzone.com
ASIO, Cubase & Digital Audio........................ http://www.steinberg.net
Cubase Users Group ........................................ http://www.groups.yahoo.com/group/
cubase/messages
Forums
Unofficial E-MU Forum ............................ http://www.productionforums.com/emu/
KVR Forum ............................................ http://www.kvr-vst.com/forum/
Driver Heaven Forum ............................ http://www.driverheaven.net/search.php?s
MIDI Addict Forum................................ http://forum.midiaddict.com/search.php
Home Recording Forum ........................ http://homerecording.com/bbs/
search.php?s=d866b60193933eb726660e7b
d90dfb27
Sound-On-Sound Forum ....................... http://sound-on-sound.com/forum/
Studio-Central Cafe Forum ................... http://studio-central.com/phpbb/search.php
Sound Card Benchmarking ................... http://audio.rightmark.org
E-MU 1616/1616M CardBus Digital Audio System
95
7 - Appendix
Internet References
Declaration of Conformity
Trade Name:
E-MU Systems
Model No.:
EM8850
EM8870
EM8871
Responsible Party:
E-MU Systems
Address:
1500 Green Hills Road,
Scotts Valley, CA 95066 U.S.A.
This device complies with Part 15 of the FCC rules. Operation is subject to the following
two conditions: (1) This device may not cause harmful interference, and (2) this device
must accept any interference received, including interference that may cause undesired
operation.
CAUTION
You are cautioned that any changes or modifications not expressly approved in this
manual could void your authority to operate this equipment.
Note:
This equipment has been tested and found to comply with the limits for a Class B
digital device, pursuant to Part 15 of the FCC Rules. These limits are designed to provide
reasonable protection against harmful interference in a residential installation. This
equipment generates, uses, and can radiate radio frequency energy and, if not installed
and used in accordance with the instructions, may cause harmful interference to radio
communications. However, there is no guarantee that interference to radio or television
reception, which can be determined by turning the equipment off and on, the user is
encouraged to try to correct the interference by one or more of the following measures:
• Reorient or relocate the receiving antenna.
• Increase the separation between the equipment and receiver.
• Connect the equipment into an outlet on a circuit different from that to which
the receiver is connected.
• Consult the dealer or an experienced radio/TV technician for help.
The supplied interface cables must be used with the equipment in order to comply with
the limits for a digital device pursuant to Subpart B of Part 15 of FCC Rules.
96
Creative Professional
7 - Appendix
Internet References
Compliance Information
United States Compliance Information
FCC Part 15 Subpart B Class B using:
CISPR 22 (1997) Class B
ANSI C63.4 (1992) method
FCC Site No.90479
Canada Compliance Information
ICES-0003 Class B using:
CISPR 22 (1997) Class B
ANSI C63.4 (1992) method
Industry of Canada File No.IC 3171-B
European Union Compliance Information
EN55024 (1998)
EN55022 (1998) Class B
EN61000-3-2 (2001)
EN61000-3-3 (1995 w/A1:98)
Australia/New Zealand Compliance Information
AS/NZS 3548(1995 w/A1 & A2:97) Class B
EN55022 (1998) Class B
Japan Compliance Information
VCCI (April 2000) Class B using:
CISPR 22(1997) Class B
VCCI Acceptance Nos. R-1233 & C-1297
Attention for the Customers in Europe
This product has been tested and found compliant with the limits set out in the EMC
Directive for using connection cables shorter than 3 meters (9.8 feet).
Notice
If static electricity or electromagnetism causes data transfer to discontinue midway
(fail), restart the application or disconnect and connect the Firewire cable again.
E-MU 1616/1616M CardBus Digital Audio System
97
7 - Appendix
Internet References
98
Creative Professional
Index
Numerics
Index
Numerics
1010 PCI Card 13
176.4kHz/192kHz Sample Rate 81
1-Band Para EQ 54
1-Band Shelf EQ 54
3-Band EQ 55
48 Volt DC Adapter 11
48 Volt Phantom Power 15, 87
4-Band EQ 56
88kHz/96kHz Sample Rate 81
A
A/D - D/A Converter Type
1616 system 91
1616M system 88
AC3 Passthrough 16
ADAT Optical
at 96kHz & 192kHz 81
input/output connector 16
AES/EBU to S/PDIF Adapter 86
Analog I/O, MicroDock 18
Appearance, improving 87
ASIO
direct monitor 32
send 30
Attack, compressor 59
Automating PowerFX 73
Auto-Wah 57
Aux Bus 38
Auxiliary Effects Assignment 44
Auxiliary Returns 44
Auxiliary Sends 38
used as extra mix busses 44
B
Background program, disabling 23
Balance Control, monitor 45
Balanced Cables 18, 85
Block Diagram, mixer 22
Bypass
effect insert 50
insert 37
send/return insert 42
E-MU 1616/1616M CardBus Digital Audio System
C
CardBus Card
description 13
installing 10
removing 13
Category
create new preset 49
delete effects 49
rename effects 49
CDs, playing 29
Chorus 58
using freq. shifter 62
Clicks & Pops, in the audio 16
Clipping Indicators 15
Clock, external 26
Comb Filter 61
Compressor 58
Connection Diagrams 17, 19
Connections
ADAT optical 16
EDI cable 11
front panel 15
rear panel 18
S/PDIF 15
Connector Types 11
Core Effects
descriptions 54
listing 53
D
Damping, high frequency 64, 69
Decay Time, lite reverb 64
Decay Time, reverb 69
Delete
folder 49
FX user preset 52
mixer strip 29
Diffusion 69
Digital Cables 85
Digital Interface, S/PDIF 15
Direct Sound Source 29
Distortion 60
Doppler, effect using Rotary 66
Drivers, installing 10
Dynamic Range 88, 91
E
Echo, creating 65
E-Delay Compensator 77
Edge, distortion 60
EDI Cable 11
EDI Connector 19
99
Index
F
Effects
1-band para EQ 54
1-band shelf EQ 54
3-band EQ 55
4-band EQ 56
auto wah 57
chorus 58
compressor 58
create new folder 49
creating robot voice 65
descriptions 54
display screen 42
distortion 60
edit 48
flanger 61
frequency shifter 62
leveling amp 63
lite reverb 64
mono delays 65
overview 47
palette 47
phase shifter 66
placing into an insert location 30
preset
create new 51
delete 52
overwrite 52
rename 52
rotary 66
selecting 48
stereo delays 68
stereo reverb 69
using in VST host application 71
vocal morpher 70
E-MU 02 CardBus Card
description 13
installing 10
E-MU Icon, in taskbar 23
Envelope, reverberation 64, 69
E-Wire 76
Exit PatchMix DSP Services 23
External Clock 26
External Sync Source 26
Extra Buffers 72
F
Factory Templates 25
Flanger 61
Frequency Shifter 62
Front Panel Connections, Audio Dock 15
FX Edit Screen 50
FX Insert Chains 48
100
G
Gain, compressor 59
Ground Loop, preventing 87
Ground Lug, turntable 18
Grounding 87
H
Headphone Output 16
Help System 23
High Frequency Damping, stereo reverb 69
High Frequency Decay Factor, lite reverb 64
High Frequency Rolloff
mono delays 65
stereo delays 68
Host Input Display 43
Host Output Display 43
Hum, in the audio 87
I
Input
display 43
level
line 18
setting 34
specs 88, 91
type
mixer strip 28
red color 28
Insert
add effect 30
add send 31
add send/return 31, 32
bypass 37, 50
delete 37
menu 31
meter 34
mixer strip 30
solo 37, 50
types 30
Interface
ADAT 16
EDI 19
MIDI 19
required cable 11
S/PDIF 15
Invert, polarity 35
J
Jitter Spec
1616 system 93
1616M system 90
Creative Professional
Index
L
L
Label, scribble strip 40
Latency, monitoring without 32
LED
green 15
red 15
Level Fader 40
Leveling Amp 63
Levels, setting input 34
LFO
flanger 61
phase shifter 66
vocal morpher 70
Limiter 59
Line Level I/O, MicroDock 18
Lite Reverb 64
Low Frequency Damping 69
Low Frequency Decay Factor, lite reverb 64
M
Main
bus 41
inserts 45
output fader 45
section 41
Master
return level 41
send level 41
volume control 45
Meter
insert 33
main output 45
setting input levels using 34
MicroDock
connecting 11
inputs/outputs 14
power switch 11
Microphone Preamps 15
MIDI
breakout cable 19
I/O jacks
MicroDock 19
I/O jacks, MicroDock 19
jacks 15
Mini-Phone Outputs 19
Mixer
block diagram 22
overview 21
strip 28
aux send 38
fader 40
insert 30
label 40
mute button 40
new 29
E-MU 1616/1616M CardBus Digital Audio System
solo button 40
type 29
viewing 21
Mixer Strip
add new 29
delete 29
type 29
Monitor
balance control 45
mix 41
mute 41
output 18
level control 45
mute 45
Monitor Output, CardBus card 13
Mono Delays 65
Mute
mixer strip 40
monitor 41
N
New
mixer strip 29
preset category 49
session 23, 24
at 176k/192k 81
user effect preset 51
Notes, Tips & Warnings 8
O
OpAmp Type
1616 system 91
1616M system 88
Optical Cables 86
Optical S/PDIF 16, 27
Output
fader, main 45
level
line 18
meters 45
monitor 45
routing display 43
section 45
P
Palette, effects 47
Pan 40
Pan Controls 28
Parametric EQ, setting up 55
PatchMix DSP, disabling 23
Peak Meters 33
Phantom Power 15
description 87
Phase Invert 35
101
Index
R
Phase Shifter 66
Phoneme 70
Physical Input Display 43
Physical Output Display 43
Physical Source 29
Pink Noise Generator 36
Playing CDs 29
Post Gain, leveling amp 63
Power Switch, microdock 11
PowerFX 71
resource availability 73
Preamp
microphone 15
turntable 18
Pre-Delay, compressor 59
Pre-Fader Aux Sends 41
Preset
create new 51
delete 52
overwrite effects 52
rename effects 52
select user 51
R
Ratio, compressor 59
Red Strip 28
Reducing Noise 87
Release, compressor 59
Render Mode 72
Reverb, envelope 64, 69
Reverberation 69
Robot Voice Effects, creating 68
Rotary, effect 66
S
S/MUX 81
S/PDIF
cables 86
inputs and outputs 15
optical 16
S/PDIF to AES/EBU Adapter 86
Safely Remove Hardware Icon 13
Sample Rate
88k, 96k, 176k & 192k 81
setting 24
Save
effect user preset 51, 52
FX Insert Chains 48
session 25
Scribble Strip 40
Send
/return insert 31, 32
bypass or solo 42
auxiliary 38
102
insert 31
Send/Return Levels 41
Session 24
creating 176k/192k 25, 81
creating new 24
templates 25
at 176k/192k 25, 81
Setting Up the E-MU Digital Audio System 9
Settings
I/O 26
input level 15
system 25
Sidechain Effects 44
routing 38
Signal Generator Insert 36
Signal Level Indicators
LEDs 15
meters 45
Signal Level, increasing 18
Sine Wave Oscillator 36
Soft Limiters 35
Software Installation 10
Solo
button 40
insert 37, 50, 51
send/return insert 42
Specifications
1616 System 91
1616M System 88
Stereo Delays 68
Stereo Reverb 69
Strip
add new 29
input type 28
mixer 28
Sync/Sample Rate Indicators 44
Synchronization Source 26
System Settings 25
T
Templates, session 25
Test Tone Insert 36
Threshold, compressor 59
Toggle Tooltips 72
Toolbar, overview 23
Trim Pot Insert 35
Troubleshooting, using test tone & meter inserts 36
TRS Plugs & Jacks 85
Turntable Inputs 18
Tutorial
Automating PowerFX 73
Making the Best Possible Recording 34
Setting up & using E-Wire 77
Setting up & using PowerFX 72
TV Screen 41, 42
Creative Professional
Index
U
U
Unbalanced Cables 85
User Preset, effect 51
V
Vocal Morpher 70
Volume Control 28
W
Wah-Wah 57
WDM Recording & Playback Behavior 83
Wet/Dry Mix, effects 50
White Noise Generator 36
Window Appearance Settings 87
Windows Media Player 29
Windows Taskbar, E-MU icon 23
X
XLR Connector 15
Z
Zero-Latency Monitoring 32
E-MU 1616/1616M CardBus Digital Audio System
103
Index
Z
104
Creative Professional
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