Index ............................................................................ 111
6Creative Professional
Page 7
ā¢
ā¢
ā¢
1- Introduction
Welcome!
Thank you for purchasing the E-MU 1616 PCI, E-MU 1616M PCI, or 1212M PCI Digital
Audio System. Your computer is about to be transformed into a powerful audio
processing workstation. Weāve designed this E-MU digital audio system to be logical,
intuitive and above all, to provide you with pristine sound quality. These systems offer
unprecedented quality and value by providing studio-class, 24-bit/192kHz multiĀchannel recording and playback to any PCI card bus equipped PC.
1- Introduction
Welcome!
1616 PCI System Components
E-MU 1616 PCI
⢠E-MU 1010 PCI Card
⢠MicroDock
⢠EDI (E-MU Digital Interface Cable)
⢠+48VDC AC Adapter
MIDI Breakout Cable
Digital Audio System Software/Driver Installation CD-ROM
Production Tools Software Bundle CD-ROM
⢠Quick Start Guide
Inputs & Outputs
(8) Channel ADAT Digital Optical Input
(8) Channel ADAT Digital Optical Output
(2) Channel S/PDIF Digital Input
(2) Channel S/PDIF Digital Output
(2) MIDI Inputs & Outputs
(4) 24-bit Balanced Line Inputs
(6) 24-bit Balanced Line Outputs
(2) Microphone/Line Preamp Inputs
(2) Turntable Preamp Inputs
(1) Stereo Headphone Output
(3) Stereo Computer Speaker Outputs
(allows 32 MIDI channels)
(with +48V phantom power)
(with RIAA equalized preamplifier)
(with volume control)
(with 1/8ā jacks to connect powered speakers)
E-MU PCI Digital Audio Systems7
Page 8
ā¢
ā¢
ā¢
ā¢
1- Introduction
Welcome!
1616M PCI System Components
E-MU 1616M PCI
⢠E-MU 1010 PCI Card
⢠MicroDockM
⢠Sync Card
⢠EDI (E-MU Digital Interface Cable)
⢠+48VDC AC Adapter
MIDI Breakout Cable
Digital Audio System Software/Driver Installation CD-ROM
Production Tools Software Bundle CD-ROM
⢠Quick Start Guide
Inputs & Outputs
(8) Channel ADAT Digital Optical Input
(8) Channel ADAT Digital Optical Output
(2) Channel S/PDIF Digital Input
(2) Channel S/PDIF Digital Output
(2) MIDI Inputs & Outputs
(allows 32 MIDI channels)
(4) 24-bit Balanced Line Inputs
(6) 24-bit Balanced Line Outputs
(2) Microphone/Line Preamp Inputs
(2) Turntable Preamp Inputs
⢠Digital Audio System Software/Driver Installation CD-ROM
Production Tools Software Bundle CD-ROM
⢠Quick Start Guide
Inputs & Outputs
(8) Channel ADAT Digital Optical Input
(8) Channel ADAT Digital Optical Output
(2) Channel S/PDIF Digital Input
(2) Channel S/PDIF Digital Output
(1) MIDI Input & Output
(allows 16 MIDI channels)
(2) 24-bit Balanced Line Inputs
(2) 24-bit Balanced Line Outputs
8Creative Professional
Page 9
All Systems Include:
The E-MU 1010 PCI Card is the heart of all three systems. Its powerful hardware DSP
processor allows you to use over 16 simultaneous hardware-based effects, which place
minimal load on your computerās CPU. The E-MU 1010 PCI Card also provides eightĀchannels of ADATĀ® optical digital input and output, as well as a S/PDIF stereo digital
input and output.
The PatchMix DSP mixer application is included in all the systems. PatchMix DSP
delivers unmatched ļ¬exibility in routing your audio between physical inputs and
outputs, virtual (ASIO/WAVE) inputs and outputs and internal hardware effects and
busesāno external mixer needed. You can add digital effects, EQs, meters, level controls
and ASIO/WAVE sends anywhere you like in the signal chain.
Because the effects and mixing are hardware-based, there is no latency when you record.
You can even record a dry signal while monitoring yourself with effects! Mixer setups
can be saved and instantly recalled for speciļ¬c purposes such as recording, mixdown,
jamming, special effect setups, playing games, watching DVDs, or general computer use.
1- Introduction
Welcome!
E-MU 1212M System
The E-MU 1212M includes the 0202 Daughter Card, which provides 2 line level,
balanced analog inputs, 2 line level, balanced analog outputs, plus MIDI input and
output. This is no-compromise audio interface, using ultra-high performance
24-bit/192kHz A/D - D/A converters to deliver an unbelievable 120dB dynamic range.
E-MU 1616 System
The E-MU 1616 includes the MicroDock, which is a half rack-space, audio interface. The
MicroDock adds the following input and output capabilities to the system: two mic/line
inputs with custom low-noise preamps, 4 balanced line level analog inputs, an RIAA
stereo turntable preamp, 6 balanced line level outputs, an assignable headphone
output, two sets of MIDI I/O ports, an additional S/PDIF optical output, and four stereo
mini phone jacks for easy connection to powered speaker systems. Of course, profesĀsional-quality, 24-bit A/D and D/A converters with automatic DC blocking are used
throughout.
E-MU 1616M System
The E-MU 1616M system includes the MicroDockM, a no compromise, mastering-grade
system, which includes all the features of the 1616 system. The 1616M system is distinĀguished by the addition of ultra-high performance 24-bit/192kHz A/D - D/A converters
which deliver an incredible 120dB of dynamic range.
The Sync Daughter Card comes standard with the 1616M system and can be purchased
as an optional upgrade for the 1616 MicroDock and 1212M systems. The Sync Card
adds Word Clock in and out for sample-synchronizing outboard digital equipment and
SMPTE longitudinal time code in/out for syncing other recording equipment. A separate
MIDI Time Code output port on the Sync Card eliminates timing problems caused by
combining MTC with MIDI performance data.
E
Note: The digital I/O
on the 1010 PCI card are
not used when you have
the MicroDock.
Youāll want to keep up with the latest software and options for your E-MU digital audio
system. You can ļ¬nd all of this, plus other helpful information, at the E-MU Website:
http://www.emu.com.
E-MU PCI Digital Audio Systems9
Page 10
1- Introduction
Welcome!
PatchMIx DSP
PatchMix DSP offers unmatched ļ¬exibility 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 speciļ¬c purposes such as
recording, mixdown, jamming, special effect setups, playing games, watching DVDs, or
general computer use.
Youāll want to keep up with the latest software and options for your E-MU digital audio
system. You can ļ¬nd 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.
fNotes provide additional information related to the topic being discussed. Often,
notes describe the interaction between the topic and some other aspect of the
system.
ETips describe applications for the topic under discussion.
ī
Warnings are especially important, since they help you avoid activities that can
cause damage to your ļ¬les, your computer or yourself.
10Creative Professional
Page 11
2 - Installation
Setting Up the Digital Audio System
There are six basic steps to installing your E-MU system:
Remove any other sound cards you have in your computer.(Once you are sure that
1.
the E-MU card works properly, your old sound card can be reinstalled if desired.)
Install the E-MU 1010 PCI card in your computer. Go there.
2.
Install the 0202 Daughter Card or Sync Daughter Card (if applicable). Go there.
3.
4.
Connect the MicroDock (if applicable).
5.
Install the PatchMix DSP software onto your computer.
Connect audio, MIDI and synchronization cables between the E-MU system and
6.
your other gear.
Notes for Installation
2 - Installation
Setting Up the Digital Audio System
⢠IF AT ANY TIME DURING THIS INSTALLATION YOU SEE NO RESPONSE:
Use the Alt-Tab feature to select other applications. One of them may be the
Microsoft Digital Signature warning. It is possible for this warning to appear
behind the installation screen.
ā¢Make sure you have the latest Windows Service Packs from Microsoft
(Windows XP - SP 1 or higher).
⢠Disable onboard sound and uninstall all other sound cards. (If you wish to try
using multiple sound cards in your system, do so after you have conļ¬rmed that
your E-MU Digital Audio System is operating normally.)
⢠InstallShield āIKernel Application Errorā on Windows XP: When installing this
software on Windows XP, you may be confronted with a ākernel errorā at the very
end of installation. This is an issue with the InstallShield program, which is what
we use to install software on your computer. Please do not be alarmed by this, as
the error is innocuous.
To read more about this error, and obtain instructions on how to avoid getting
the message, please visit this website:
http://support.installshield.com/kb/view.asp?articleid=q108020
ā¢Multiple Digital Audio System sound cards are not supported.
E-MU PCI Digital Audio Systems11
Page 12
2 - Installation
Setting Up the Digital Audio System
Please read the following sections as they apply to your system as you install the E-MU
1010, paying special attention to the various warnings they include.
Prior to installing the hardware, take a few moments to write down the 18-digit serial
number, which is located on the back of the box and on the 1010 PCI Card. This
number can help EMU Customer Service troubleshoot any problems you may
encounterāby writing the number down now, youāll avoid having to open your
computer to ļ¬nd it later on.
Safety First!
⢠To avoid possible permanent damage to your hardware, make sure that all connecĀtions are made with the host computerās power off. Unplug the computerās
power cable to make sure that the computer is not in sleep mode.
⢠Take care to avoid static damage to any components of your system. Internal
computer surfaces, the E-MU 1010 PCI board and the interfaces are susceptible to
electrostatic discharge, commonly known as āstatic.ā Electrostatic discharge can
damage or destroy electronic devices. Here are some procedures you can follow
when handling electronic devices in order to minimize the possibility of causing
electrostatic damage:
As you install
ī
hardware components,
observe the following
general precautions to
avoid damage to your
equipment and yourself.
⢠Avoid any unnecessary movement, such as scufļ¬ng your feet when handling
electronic devices, since most movement can generate additional charges of static
electricity.
⢠Minimize the handling of the PCI card. Keep it in its static-free package until needed.
Transport or store the board only in its protective package.
⢠When handling a PCI card, avoid touching its connector pins. Try to handle the
board by its edges only.
⢠Before installing a PCI card into your computer, you should be grounded. Use a
ground strap to discharge any static electric charge built up on your body. The
ground strap attaches to your wrist and any unpainted metal surface within your
computer. If you donāt have a ground strap, you can ground yourself by touching
the metal case of another piece of grounded equipment.
⢠Before connecting a cable to your interface or between PCI cards, touch the
connector sleeve of the cable to the sleeve of the jack to which youāll be connecting
the cable in order to discharge any static build-up.
Connector Types
These connector types are used to connect the E-MU 1010 hardware components. They
will be referred to by the name shown in the ļ¬rst column of the following chart:
NameDescriptionConnects
Card/ExternalCAT5 Connector1010 PCI card and MicroDock
S/PDIF InRCA ConnectorS/PDIF digital audio devices
S/PDIF OutRCA ConnectorS/PDIF digital audio devices
ADAT Optical Out TOSLINK Optical Connector ADAT digital audio devices (or S/PDIF)
ī
Warning: Please verify that all cables are connected only to the proper components
before powering up your system.
12Creative Professional
Page 13
Installing the E-MU 1010 PCI Card
Note: This installation is very simple but if you are not familiar with the installation
of computer peripherals and add-in boards, please contact your authorized E-MU
Systems dealer or an approved computer service center to arrange for the installation.
To install the 1010 PCI card into your computer
ī
1.
Make sure that the power switch on your computer is off.
IMPORTANT: Unplug the power cord from the wall outlet!
2.
Touch a metal plate on your computer to ground yourself and to discharge any
static electricity.
Follow the computer manufacturerās recommended procedure for opening the case.
3.
4.
Remove the metal brackets from two adjacent PCI slots. If you have the E-MU 1820
system (non-M) you only need to remove the bracket from a single PCI slot. Put the
screw(s) aside for use later. See ļ¬gure 1 below.
Figure 1Figure 2
2 - Installation
Installing the E-MU 1010 PCI Card
Note: Some
E
computer cases donāt use
screws to secure PCI
cards. In this case, follow
the instructions that
came with your
computer.
PCI Slots
(may not be present
ISA Slots
on your computer)
Align the E-MU 1010 PCI card with the slot and press gently but ļ¬rmly down into
5.
the slot as shown in ļ¬gure 2.
Do not force the E-MU 1010 card into the slot. Make sure that the gold ļ¬nger
6.
connector of the card is aligned with the PCI bus connector on the motherboard
before you insert the card into the PCI slot. If it doesnāt ļ¬t properly, gently remove it
and try again.
Secure the card into the slot using one of the screws you placed aside earlier.
7.
E-MU PCI Digital Audio Systems13
Page 14
2 - Installation
Install the Sync Daughter Card or 0202 Daughter Card
Install the Sync Daughter Card or 0202 Daughter
Card
E-MU 1616 - If youāre planning to use Word Clock. MIDI Time Code or SMPTE
sync, unwrap the Sync Daughter Card and get ready to install it. If you donāt need
these options or donāt have an empty PCI slot, you can skip these next few steps.
ā¢E-MU 0202 - Unwrap the 0202 Daughter Card and get ready to install it.
1.
Connect the ribbon cable provided with the kit between the E-MU 1010 card and
the 0202 Daughter card or Sync Daughter Card as shown in ļ¬gures 3 & 4. The cables
are keyed so they cannot be incorrectly inserted. Seat the connectors ļ¬rmly in the
sockets and arrange the cables neatly.
2.
Align the Sync Daughter Card or the 0202 Daughter Card with the slot and press
gently but ļ¬rmly down into the slot as shown in ļ¬gure 2 on the preceding page.
3.
Do not force the E-MU Card into the slot. Make sure that the tab at the rear of the
card is aligned with the PCI bus connector on the motherboard before you insert
the card into the PCI slot. If it doesnāt ļ¬t properly, gently remove it and try again.
Secure the card into the slot using one of the screws you placed aside earlier.
4.
ā¢
Figure 3Figure 4
Sync Daughter
Card
After all components have been installed and securely fastened, close the computer
5.
0202 Daughter
Card
case.
6.
Connect the supplied network-type cable from the 10 BaseT jack on the E-MU 1010
PCI card labeled āEDIā to the matching connector labeled āEDIā on the MicroDock.
The cable supplied with the MicroDock is specially shielded to prevent unwanted
RF emissions.
Plug the power cord back into the wall outlet and turn on your computer.
7.
WARNING: E-MU 0202 & MicroDock
If you have both the E-MU 0202 I/O card and the MicroDock, DO NOT connect both to
the E-MU 1010 PCI card.
ī
CAUTION: Do not
connect the supplied
CAT5 cable to the
Ethernet or network
connector on your
computer. Doing so may
result in permanent
damage to either your
computer, the E-MU 1010
or both.
14Creative Professional
Page 15
Connecting the MicroDock
1.
Connect the supplied EDI cable between the 1010 PCI Card and the MicroDock.
Connect the supplied +48 volt DC adapter to the +48VDC jack on the rear of the
2.
MicroDock. See the diagram below.
3.
Connect your audio inputs and outputs to the MicroDock as shown on page
Turn the MicroDock on by turning the Headphone Volume control.
4.
+48V DC Adapter
48
VDC
+
-
23.
2 - Installation
Connecting the MicroDock
The 1616 and 1616
ī
MicroDocks cannot be
used with older 1010 PCI
cards identiļ¬ed by the
1394 FireWire port.
m
EDI
The Headphone
Volume Control is
the Power Switch.
1010 PCI Card
Warning:
ī
The MicroDock has been designed to use readily available and
inexpensive standard computer system cables. This makes it easy for you to ļ¬nd
replacement cables if your original cable becomes damaged or lost. However, because
these standard cables types are used for other purposes, you must use caution to avoid
connecting the cables incorrectly. DO NOT connect the supplied EDI cable to the
Ethernet or network connector on your computer. Doing so may result in permanent
damage to either your computer, the E-MU 1010 PCI card, or the MicroDock.
E-MU PCI Digital Audio Systems15
Page 16
2 - Installation
Software Installation
Software Installation
Installing the E-MU 1010 Drivers
The ļ¬rst time you restart your PC after installing the E-MU 1010 PCI card, you will need
to install the PatchMix DSP software and E-MU 1010 PCI card drivers.
Windows XP
The software is not compatible with other versions of Windows.
1.
After you have installed your Digital Audio System, turn on your computer.
Windows automatically detects the Digital Audio System and searches for device
drivers.
2.
When prompted for the audio drivers, click the
3.
Insert the E-MU software Installation CD into your CD-ROM drive. If Windows
AutoPlay mode is enabled for your CD-ROM drive, the CD starts running automatiĀcally. If not, from your Windows desktop, click
(replace
and double-click
4. The installation splash screen appears. Follow the instructions on the screen to
d:\
with the drive letter of your CD-ROM drive). You can also open the CD
Setup.exe
.
complete the installation.
5. Choose āContinue Anywayā when you encounter the āWindows Logo Testingā
warning screen. See the note below for more information.
6. When prompted, restart your computer.
Cancel
button.
Start->Run
and type
d:\setup.exe
E
Serial Number -
During the registration
process, you will be asked
to enter your 18-digit
serial number. The serial
number is located on the
back of the box and on
the 1010 PCI Card.
Uninstalling all Audio Drivers and Applications
At times you may need to uninstall or reinstall some or all of the audio card's applicaĀtions and device drivers to correct problems, change conļ¬gurations, or upgrade
outdated drivers or applications. Before you begin, close all audio card applications.
Applications still running during the uninstallation will not be removed.
1. Click Start -> Settings -> Control Panel.
2. Double-click the Add/Remove Programs icon.
3. Click the Install/Uninstall tab (or Change or Remove Programs button).
4. Select the E-MU driver/application entries and then click the Add/Remove (or
Change/Remove) button.
5. In the InstallShield Wizard dialog box, select the Remove option.
6. Click the Yes button. Restart your computer when prompted.
7. You may now re-install existing or updated E-MU 1010 PCI card device drivers or
applications.
Note About Windows Logo Testing
When you install the Digital Audio System drivers, you will see a dialog box that
informs you that the driver has not passed Windows Logo testing.
The Digital Audio System drivers are not signed because the driver does not support
some of the consumer audio features that the Microsoft driver signing program requires,
most notably Digital Rights Management.
However, the Digital Audio System drivers have been rigorously tested using the same
test procedures that a signed driver requires, and it passes in all important categories,
including those that measure the relative stability of the driver. So, it is perfectly safe to
install these drivers on your computer.
16Creative Professional
Page 17
3 - PCI Card & Interfaces
The E-MU 1010 PCI Card
The E-MU 1010 PCI card is the heart of the system and contains E-MUās powerful E-DSP
chip. The powerful hardware DSP on this card leaves more power free on your CPU for
additional software plug-ins and other tasks.
Important
When the MicroDock is connected to the 1010 PCI card, the digital I/O on the PCI
card is disabled. Use the digital I/O on the MicroDock.
Connections
EDI Connector
Connects to the MicroDock using the supplied EDI
cable. This cable provides a a two-way data link
EDI
Connects to
MicroDock
via EDI Cable
S/PDIF
In/Out
between the E-MU 1010 and the MicroDock as well as
supplying power to the MicroDock.
S/PDIF Digital Audio Input & Output
RCA phono jacks are standard connectors used for
S/PDIF (Sony/Philips Digital InterFace) connections.
Each jack carries two channels of digital audio. The
E-MU 1010 receives digital audio data with word
lengths of up to 24-bits. Data is always transmitted at
24-bits.
3 - PCI Card & Interfaces
The E-MU 1010 PCI Card
S/PDIF digital I/O can be used for the reception and/
or transmission of digital data from external digital
devices such as a DAT external analog-to-digital
ADAT
or S/PDIF
Optical
In/Out
converter or an external signal processor equipped
with digital inputs and outputs.
The S/PDIF out can be conļ¬gured in either ProfesĀsional or Consumer mode in the Session Settings
menu. The 1010 PCI card can also send and receive
AES/EBU digital audio through the use of a cable
adapter. See AES/EBU to S/PDIF Cable Adapter for
details.
The S/PDIF input and outputs are usable at the
44.1kHz, 48kHz 88.2kHz and 96kHz sample rates,
but are disabled for 176.4kHz and 192kHz. The word
clock contained in the input data stream can be used
as a word clock source. See System Settings.
ADAT Optical Digital Input & Output
The ADAT optical connectors transmit and receive 8 channels of 24-bit audio using the
ADAT type 1 & 2 formats. The word clock contained in the input data stream can be
used as a word clock source. See System Settings. Optical connections have certain
advantages such as immunity to electrical interference and ground loops. Make sure to
use high quality glass ļ¬ber light pipes for connections longer than 1.5 meters.
ī Important: When
using any type of digital
I/O such as S/PDIF or
ADAT, you MUST sample
sync the two devices or
clicks and pops in the
audio will result.
E-MU PCI Digital Audio Systems17
Page 18
3 - PCI Card & Interfaces
The 0202 Daughter Card
At the 96kHz or 192kHz sample rates, the industry standard S/MUX interleaving
scheme is used for ADAT input and output. S/MUX uses additional ADAT channels to
achieve the required bandwidth. See the chart below or go here for additional inforĀmation.
Sample RateNumber of Audio Channels
44kHz/48kHz8 channels of 24-bit audio
88.2kHz/96kHz4 channels of 24-bit audio, using S/MUX standard
176.4kHz/192kHz2 channels of 24-bit audio, using S/MUX standard
The 0202 Daughter Card
The 0202 Daughter card is the companion card for E-MU 1010 systems which donāt
include the MicroDock. The 0202 Daughter card provides one pair of 24-bit balanced
analog inputs and one pair of 24-bit balanced analog outputs, plus MIDI in and out.
Connections
Analog Inputs and Outputs
The 0202 Daughter Card provides two balanced,
analog inputs and two balanced, line level analog
outputs. The inputs can be connected to any line level
Left / Right
Line Inputs
stereo signal from keyboards, CD-players, cassette
decks, etc. The analog inputs are assigned to a mixer
strip in the mixer application.
The outputs can feed any line level input such as a
Left / Right
Line Outputs
mixing board, the auxiliary input on your stereo or a
set of powered speakers. The line outputs are not
designed to drive headphones directly. Connect the
line outputs to a stereo receiver or mixer with a
headphone jack to obtain the proper current drive.
MIDI
In/Out
Either TRS (tip-ring-sleeve) balanced or TS unbalanced
cables can be used. Balanced cables provide better
noise immunity and +6dB higher signal level. The
output line level can be set to accommodate the
consumer -10dBV standard, or the pro audio +4 dBu
standard in the I/O screen of the Session Settings
dialog box. See I/O Settings.
MIDI In/Out
The MIDI input and output port can be assigned in your speciļ¬c MIDI application.
Connect the MIDI adapter cable that came with your 0202 Daughter card to the miniĀDIN connectors on the card. The adapter cables convert the mini-DIN to standard DIN
connectors used on most keyboards and synthesizers. Connect MIDI Out to the MIDI In
port of your synthesizer and MIDI Out of your synth to MIDI In of the 0202 Daughter
Card.
18Creative Professional
Page 19
The MicroDock
The MicroDock connects to the E-MU 1010 PCI card via the EDI cable.
The MicroDock provides (4) balanced analog inputs, (2) microphone preamp inputs,
(6) balanced line-level analog outputs, (3) stereo 1/8ā outputs for connecting powered
computer speakers, (2) MIDI inputs, (2) MIDI outputs, a stereo headphone output, and
a RIAA equalized turntable preamp section which is ānormalledā into line input 2L and
2R, 8 channels of ADAT digital input/output, and stereo S/PDIF digital input/output.
3 - PCI Card & Interfaces
The MicroDock
f The MicroDock is
completely āhot
pluggableāā Itās OK to
plug or unplug the
MicroDock while the
computer is turned on.
Out
Line
A
In
Mic
Clip
SL
-15
Line -
0
Mic -
1L
1R
1L
1R
Line
B
-3
-6
-12
-20
+50
+65
2L
2L
Mic
Clip
-3
-6
SL
-12
-20
-15
0
Phono
2R
2R
2L
3L
48V
+50
+65
2R
Gnd
3R
S/PDIF
In
MIDI Cable
Out
2
1
The inputs are conļ¬gured as follows:
(2)mono microphone/line inputs (2 inputs)
(2)stereo pairs of line level inputs (4 inputs)
(1)stereo pair of S/PDIF/AES digital inputs (2 inputs)
(4)stereo pairs of ADAT channels on the ADAT optical input (8 inputs)
Out
ī Itās a good idea to
mute MicroDock inputs 2
in the PatchMix DSP
Off
mixer when nothing is
plugged in, since the
turntable preamp has a
very high gain (60dB)
and could contribute
48
VDC
+
-
3
EDI
extra noise to your mix/
monitor bus.
(1)RIAA equalized turntable preamp input allows you to connect a turntable without using
an expensive external preamp.
Note: These inputs are automatically disconnected
when plugs are inserted into inputs 2L & 2R.
(2)MIDI input ports using the supplied breakout cable
The outputs are conļ¬gured as:
(3)stereo pairs of line level outputs
(1)stereo pair driving a stereo headphone jack
(1)stereo pair of S/PDIF/AES digital outputs
(4)stereo pairs of ADAT channels on the ADAT optical output
(3)stereo 1/8ā computer speaker outputs. These outputs carry the same signals as the 3
stereo line level outputs and are provided as a convenience for connecting computer or
powered speaker systems.
(2)MIDI output ports using the supplied breakout cable
E-MU PCI Digital Audio Systems19
(Share the same routing as Line Outs 1L/1R)
Page 20
3 - PCI Card & Interfaces
The MicroDock
Front Panel Connections
Preamp Section
The front panel mono Mic/Line inputs A & B can be used as balanced microphone
inputs, hi-Z guitar pickup inputs, or line level inputs. The Neutrik combination jack
accepts microphones using a standard XLR connector or line level/hi-Z inputs (such as
an electric guitar) using a standard 1/4 inch TRS/TS connector.
Each preamp has a level control which sets the preamp gain from 0dB to +65dB for the
XLR input and from -15dB to +50dB for the Hi-Z line input. The line markings around
the knobs are calibrated in 10dB increments. The heavy hash marks on the gain controls
indicate unity analog gain to the converter inputs (~5dBV input = 0dBFS output).
A phantom power switch enables +48 volt phantom power supplied to both microĀphones. A red LED illuminates to indicate phantom power is enabled. The audio mutes
for a second when phantom power is turned on. After turning phantom power off, wait
two full minutes before recording to allow the DC bias to drain. See Phantom Power for
additional information.
Each microphone input has its own input level meters and clipping indicators. The LED
meters indicate signal presence. Adjust the input gain so that the yellow LEDs are illumiĀnated. The red Clip LED indicates that the gain is set too high and the signal is clipping
the input. These LEDs monitor the signal directly at the analog-to-digital converters and
before any processing by the rest of the system. When setting the levels for signals being
sent into the MicroDock, the red clip indicator should never ļ¬ash.
ī Phantom Power
Caution: Some
microphones (notably
ribbon types) cannot
tolerate phantom power
and may be damaged.
Check the speciļ¬cations
and requirements of your
microphone before using
phantom power.
S/PDIF Digital Audio Input & Output
RCA phono jacks are standard connectors used for coaxial S/PDIF (Sony/Philips Digital
InterFace) connections. Each jack carries two channels of digital audio. The MicroDock
sends or receives digital audio data at 44.1k, 48k, 88.2k, 96k, 176.4k or 192k sample
rates. Data is always transmitted at 24-bits, but lower word widths can be read. The word
clock contained in the input data stream can be used as a word clock source. See System
Settings.
S/PDIF digital I/O can be used for the reception and/ or transmission of digital data
from external digital devices such as a DAT, external analog-to-digital converter or an
external signal processor equipped with digital inputs and outputs.
The S/PDIF out can be conļ¬gured 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.
20Creative Professional
Page 21
ADAT Optical Digital Input & Output
The ADAT optical connectors transmit and receive 8 channels of 24-bit audio using the
ADAT type 1 & 2 formats. The word clock contained in the input data stream can be
used as a word clock source. See System Settings. Optical connections have certain
advantages such as immunity to electrical interference and ground loops. Make sure to
use high quality glass ļ¬ber light pipes for connections longer than 1.5 meters.
At the 88.2k, 96k, 176.4k or 192k sample rates, the industry standard S/MUX interĀleaving scheme is used for ADAT input and output. S/MUX uses additional ADAT
channels to gain additional bandwidth on the existing interface. See the chart below or
go here for additional information.
Sample RateNumber of Audio Channels
44kHz/48kHz8 channels of 24-bit audio
88kHz or 96kHz4 channels of 24-bit audio, using S/MUX standard interleaving
176kHz or 192kHz2 channels of 24-bit audio, using S/MUX standard interleaving
The ADAT inputs and outputs can be conļ¬gured in the System Settings (page 31) to
send and receive S./PDIF optical data at 44.1k, 48k, 88.2k, or 96k sample rates.
S/PDIF Optical is not supported at 176.4k or 196k.
3 - PCI Card & Interfaces
The MicroDock
ī Important: When
using any type of digital
I/O such as S/PDIF or
ADAT, you MUST sample
sync the two devices or
clicks and pops in the
audio will result.
Headphone Output & Volume Control
The headphone output drives standard stereo headphones and the adjacent volume
control sets the listening level. The headphone ampliļ¬er can drive headphones with
impedance as low as 24 ohms. The headphone output uses a high-current version of the
high-quality output ampliļ¬ers 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.
4 balanced 24-bit, line-level, analog inputs are provided (1-2). These can be used to
input any line level signal from keyboards, CD-players, cassette decks, etc. The analog
inputs are assigned to mixer strips in the mixer application. The line level inputs can be
set to accommodate the consumer -10dBV standard, or the pro audio +4 dBu standard
in the I/O screen of the Session Settings dialog box. See I/O Settings.
The maximum input level is 18dBV (=20.2dBu).
Either TRS balanced or TS unbalanced cables can be used. See page 96 for additional
information about unbalanced cables and connectors. The line-level inputs are all
servo-balanced, enabling them to convert unbalanced signals to balanced signals
internally to reduce noise.
Turntable Inputs & Ground Lug
The RCA turntable inputs feed an RIAA equalized preamp designed for moving magnet
type phono cartridges with 60 dB of gain. Connect the ground lead from your turntable
to the ground lug to prevent hum.
The turntable inputs share line level inputs 2L and 2R. Inserting a plug into Line Input 2
disconnects the turntable preamp from that channel. Do NOT leave your turntable
connected when using inputs 2L and 2R, since this can cause a ground loop.
Important: Do NOT plug in line level signals to the turntable inputs. The turntable
inputs are designed to accept the extremely low-level signal from a phonograph
cartridge. Use RCA to 1/4ā adapters to connect line level signals to the line level analog
inputs.
Line Level Analog Outputs
Six balanced 24-bit, line-level, analog outputs are provided (1-3). Output pair 1 is desigĀnated as the Monitor Output and is fed by the monitor bus of the PatchMix DSP mixer
application. We suggest that you plug your speakers in here. Special anti-pop circuitry
mutes the analog outputs when power is turned on or off.
Like the analog line inputs, either TRS balanced or TS unbalanced cables can be used.
Balanced cables provide better noise immunity and +6dB higher signal level. The output
line level can be set to accommodate the consumer -10dBV standard, or the pro audio
+4 dBu standard in the I/O screen of the Session Settings dialog box. See I/O Settings.
The maximum input and output line levels are matched when the input and output
settings are set to the same mode (pro or consumer) in the I/O preferences screen.
ī Itās 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.
ī Balanced Cables:
You should ONLY use
balanced (TRS) cables if
BOTH pieces of
equipment use balanced
connections. Connecting
balanced cables between
balanced outputs and
unbalanced inputs can
actually increase noise
and introduce hum.
E-MU PCI Digital Audio Systems23
Page 24
3 - PCI Card & Interfaces
table
The MicroDock
Computer Speaker Analog Outputs
These stereo mini-phone (3.5mm) jacks duplicate line level outputs 1-3 with a lower
output level to accommodate consumer speakers. These line level outputs are designed
to interface easily with powered speakers.
Computer Speaker OutputDuplicates Line Level Output
1 L/RTip = 1L Ring = 1R
2 L/RTip = 2L Ring = 2R
3 L/RTip = 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 āSounds and Audio Devicesā.
There are two completely independent sets of MIDI input and output ports on the
MicroDock, which can be assigned in your speciļ¬c MIDI applications.
Connect the MIDI breakout cable to the D-connector on the MicroDock. Connect MIDI
Out to the MIDI In port of your synthesizer and MIDI Out of your synth to MIDI In of
the MicroDock MIDI cable.
EDI Connector (Card)
Connects the MicroDock to the E-MU 1010 PCI card using a CAT5-type computer cable.
The cable supplied with the MicroDock is specially shielded to prevent unwanted RF
emissions.
Basic
Connections
Audio
from
Synthesizer
In
Out
Audio
to
Monitors
MIDI Synthesizer
1L
1R
1L
1R
Mixer
Speakers
**
2R
2L
2L
2R
&
MIDI In
Out
MIDI 1
MIDI Out
Phono
2L
2R
3R
3L
Gnd
MIDI Cable
Out
2
1
In
48
VDC
+
3
EDI
Connect
Desktop
Speakers to
1/8" jacks
e
r
o
e
t
S
Turn
-
AC Adapter
1010 PCI
Card
Powered
Desktop
Speakers
* Note: Line Inputs 2L/2R and Phono 2L/2R cannot be used at the same time.
24Creative Professional
Page 25
5.1 Surround Speaker Connections
3 - PCI Card & Interfaces
The MicroDock
Center
Left
Front
Phono
2L
2R
1L
In
Out
1L
2L
1R
1R
2L
2R
2R
3R
3L
Gnd
MIDI Cable
Out
2
1
3
Left
Rear
FrontRear Ctr/Sub
Sub-Woofer
(with built-in power amps)
The 1/8ā stereo jacks make it easy to connect to powered surround sound speakers. Only
three stereo cables are necessary with many speaker systems (see above). The 1/8ā jacks
duplicate the 1/4ā outputs.
48
VDC
+
-
EDI
Right
Front
Right
Rear
You can connect the 1/8ā stereo jacks to your surround speakers and connect the 1/4ā
outputs to your other gear for music creation. When you want to monitor in surround,
simply open the 5.1 Session and turn on your surround speakers.
The chart below shows how to connect the outputs for 5.1 surround sound playback.
Multichannel WAVE to Surround Sound Speaker Channels
E-DSP WAVE 1/2Front Left / Front Right1L = FL 1R = FR1 (Tip = FL Ring = FR)
E-DSP WAVE 3/4Center / Subwoofer3L = C 3R = Sub3 (Tip = C Ring = Sub)
E-DSP WAVE 5/6Rear Left / Rear Right2L = RL 2R = RR2 (Tip = RL Ring = RR)
E-DSP WAVE 7/8Side Left / Side Right N/AN/A
E-MU PCI Digital Audio Systems25
Page 26
3 - PCI Card & Interfaces
The MicroDock
26Creative Professional
Page 27
4 - The PatchMix DSP Mixer
PatchMix DSP
The PatchMix DSP Mixer is a virtual console which performs all of the functions of a
typical hardware mixer and a multi-point patch bay. With PatchMix, you may not even
need a hardware mixer. PatchMix DSP performs many audio operations such as ASIO/
WAVE routing, volume control, stereo panning, equalization, effect processing, effect
send/return routing, main mix and monitor control and allows you to store and recall
these āSessionsā at will.
ī To Invoke the PatchMix DSP Mixer
1.
Left-click once on the E-MU icon on the Windows System Tray. The PatchMix
DSP mixer window appears.
Overview of the Mixer
Add New
Strip
Physical Input Strips
ASIO Input Strip
Toolbar
4 - The PatchMix DSP Mixer
PatchMix DSP
f Click on the buttons
and knobs in the mixer
screen below to jump to
the description of the
control.
Display
Select
Buttons
Delete
Strip
Channel
Insert
Section
Pan
Controls
Aux
Sends
Volume
Fader
Solo/Mute
Buttons
āTVā
Screen
Aux
Effects
Section
Sync/
Sample
Rate
Indicators
Monitor
User
Deļ¬nable
Scribble Strip
E-MU PCI Digital Audio Systems27
Controls Windows Source Audio
(Direct Sound, Windows Media, etc.)
WAVE Strip
Main
Inserts
Current
Session
Name
Main Mix
Output Volume
& Meters
Volume/Balance
/Mute Controls
Page 28
4 - The PatchMix DSP Mixer
Overview of the Mixer
Mixer Window
The Mixer consists of four main sections.
Application Toolbar Lets you manage sessions and show/hide the various views.
Main SectionControls all the main levels, aux buses, and their inserts. This section
also has a āTVā which shows parameters for the currently selected
effect and the input/output patching. It also shows the sessionās
current sample rate and whether itās set to internal or external clock.
Mixer StripsThis 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 PaletteThis 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 simpliļ¬ed diagram of the mixer is shown below.
Input
Post-Fader Strip
Insert
Section
Panning
Input
Pre-Fader Strip
Insert
Section
Fader
MUTE
Aux 1
Aux
Bus 1
Aux 1
Send
Amount
Aux
Effects
Insert
Section
Aux 2
Aux
Bus 2
Aux 2
Send
Amount
Insert
Section
Fader
MUTE
Main Bus
Return
Amount
Return
Amount
Mixer Block Diagram
Meter
Main Bus
Effects
Insert
Section
Main
Level
Monitor
Out
MUTE
Monitor
Level
Main
Out
Output 1L/1R
& Headphones
Pre Fader or Post Fader
When creating a new Mixer Strip, you have the option for the Aux Sends to be placed
Post Fader (both Aux Sends come after the channel fader) or Pre Fader (both Aux Sends
come before the channel fader). The Pre-fader option allows you to use either Aux Send
as another mix bus, which is unaffected by the channel fader. More Information.
28Creative Professional
Page 29
E-MU Icon in the Windows Taskbar
Right-clicking on the E-MU icon in the Windows taskbar calls the following window.
Right-Click Here
Opens the PatchMix DSP Mixer.
Calls the PatchMix DSP help system.
Disables the splash screen that appears at
boot-up.
When unchecked, FX are not loaded until
needed, resulting in faster computer boot.
Restores the default PatchMix DSP and
driver settings.
Closes the PatchMix DSP background
program, disabling use of all audio I/O
from the E-MU hardware. Open the PatchĀMix DSP application to start audio again.
4 - The PatchMix DSP Mixer
E-MU Icon in the Windows Taskbar
f Restore Defaults:
Always try this option ļ¬rst
if PatchMix is crashing or if
you are having any other
strange audio problems.
The Toolbar
New
Session
Open
Session
New Session
Open SessionCalls up the standard āOpenā dialog box, allowing you to open a
Save SessionCalls up the standard āSaveā or āSave Asā¦ā dialog boxes, allowing
Show/Hide EffectsToggle button that shows or hides the FX palette.
Session SettingsCalls up the Sessions Settings window. Session Settings.
Save
Session
āAboutā
PatchMix DSP
Show/Hide
Effects
Session
Settings
Global
Prefs
Calls up the āNew Sessionā dialog box. New Session.
saved Session.
you to save the current Session.
f Click the buttons in
the toolbar to learn about
their function.
Global PreferencesCalls up the Global Preferences window.
About PatchMix DSP Right-Click on the E-MU logo to view the āAbout PatchMix DSPā
screen, which provides the software and ļ¬rmware version
numbers and other information.
E-MU PCI Digital Audio System29
Page 30
4 - The PatchMix DSP Mixer
The Session
The Session
The current state of the PatchMix DSP mixer (fader settings, effects routingsā¦everyĀthing!) can be saved as a Session. Whenever you create or modify a mixer setup, all you
have to do is Save it to be able to recall it at a later time.
Before you begin using PatchMix DSP, you need to set it up to be compatible with the
other software applications you may be running. The most important consideration is
your system sample rate. PatchMix DSP and any applications or other digital gear you
are using must be set to the same sample rate. PatchMix DSP can run at 44.1kHz,
48kHz, 88kHz, 96kHz, 176.4 kHz or 192kHz, but its complete set of features are only
available at 44.1kHz or 48kHz. See Chapter 6 - Using High Sample Rates for details.
Once the sample rate is set, you can only easily switch between 44.1k and 48k. You
cannot switch between 44/48k and 88k/96k/176k/192k. With a change to these high
sample rates, you must start a new session.
You can also set up an external sync source, thereby obtaining the sample rate from
some other device or application. External sync can be obtained from the ADAT input or
S/PDIF input. If the session is set at 44.1kHz or 48kHz and the external source is
coming in at a higher rate (such as 96k), the Sync Indicator will be extinguished (off),
but PatchMix will attempt to receive the external data. The two units are NOT sample
locked however, and you should correct this condition to avoid intermittent clicks in the
audio. Always check for the presence of the LOCKED indicator whenever you are
using a digital interface.
PatchMix DSP comes with several session templates to choose from so when you create
a new session you can either create a āblankā 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 conļ¬gĀurable.This allows you to create only those strips you need up to a maximum number
determined by available DSP resources and available inputs.
ī Important: When
using any form of digital
input, you MUST
synchronize the Digital
Audio System to the
external digital device
(S/PDIF/ADAT).
New Session
You create a new session by clicking the āNew Sessionā button in the PatchMix DSP
main Toolbar. The following dialog box appears.
Select a Template or new
Session at the desired
sample rate
Session Description
Add your own comment
or note about the Session
Check this if you want to
edit the New Session.
30Creative Professional
Page 31
You can now select one of the factory template sessions. The factory templates are preĀprogrammed with speciļ¬c 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
āSession Templatesā folder (Program Files\Creative Professional\E-MU PatchMix
DSP\Session Templates).
There is also a Comment area that you can use to give yourself some clue as to what you
were thinking when you created the session.
Selecting a Session at 176.4kHZ or 192kHz
When operating at 176.4k or 192k sample rates, the number of I/O channels are slightly
reduced. At these high sample rates you must select one of three types of sessions each
containing a different I/O conļ¬guration. Please see page
87 for details.
Open Session
To Open a saved session, click on the Open Session button. A dialog box appears
allowing you to choose one of your saved Sessions to open. Choose one of your saved
sessions and click on the Open button.
4 - The PatchMix DSP Mixer
The Session
Save Session
To Save a session, click on the Save Session button. A Save dialog box appears allowing
you to choose a location in which to save the current Session. The āMy Sessionsā folder
is chosen by default.
Get in the habit of saving the session whenever you have created a special mixer setup.
This will make your life much easier as you can recall a setup for many different audio
modes such as: recording, mixing, special ASIO routings, etc.
Session Settings
System Settings
Pressing the Session Settings button on the toolbar brings up the System Settings
window shown below. Click the tabs to select System or I/O options.
E-MU PCI Digital Audio System31
Page 32
4 - The PatchMix DSP Mixer
The Session
The System Settings include the following:
⢠Internal/External ClockSelects between internal or external word clock source
as the master clock source for the system
⢠Sample RateSelects the sample rate when using internal clock.
Your choices are: 44.1kHz, 48kHz, 88.2kHz, 96kHz,
176.4kHz, 192kHz.
⢠External Clock Source
(ext. clock only)
Select from: ADAT, or S/PDIF as an external sample clock
source.
Using External Clock
Whenever you are using any digital I/O such as ADAT or S/PDIF, one of the digital
devices MUST supply the master clock to the others. This master clock runs at the system
sample rate and can be embedded into a data stream such as S/PDIF or ADAT.
Common symptoms of unsynced digital audio include, random clicks or pops in the
audio or failure of the digital stream to be recognized. Always check for the presence of
the āLOCKEDā 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 āLOCKEDā indicatorwill 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 āCables - balanced or unbalanced?ā in the Appendix for more
information.
E Note: if set to
āExternalā without an
external clock present,
PatchMix DSP defaults to
the internal 48kHz clock
rate.
Comparison of -10dBV & +4dBu Signal Levels
Consumer
Clipping -->
Headroom
+ 6 dBV
+ 2 dBV
{
-10 dBV
0 dBV = 1V RMS 0dBu = .777V RMS
An input setting of -10 is compatible with consumer audio gear and works best with low
level signals. (-10dBV is approximately 12dB lower than +4dBu.) Choose the setting that
allows you to send or receive a full scale signal without clipping.
Setting correct input and output levels is important! You can measure the level of an
input by inserting a meter into the ļ¬rst effect location in the strip. Adjust your external
equipment outputs for the optimum signal level. See āTo Set the Input Levels of a Stripā
for details.
Professional
(balanced)(unbalanced)
+20 dBu
=
+8 dBu
=
+4 dBu
=
-8 dBu
<-- Clipping
Headroom
}
f Input too weak?
Use -10 Input setting.
Output too weak?
Use +4 Output setting
32Creative Professional
Page 33
4 - The PatchMix DSP Mixer
The Session
Input Level
Settings
Optical
Input
Select
Mic Soft
Limiting
On/Off
⢠Inputs +4 or -10Selects between Consumer level (-10dBV) or
Professional level (+4dBu) inputs.
(Use the -10dBV setting if your input is too weak.)
⢠Outputs +4 or -10Selects between Consumer level (-10dBV) or
Professional level (+4dBu) outputs.
(The +4 dBu setting outputs a hotter level.)
⢠Optical Input SelectSelects between ADAT or optical S/PDIF for the MicroDock
ADAT Input. The coaxial S/PDIF input is disabled when
S/PDIF optical is selected.
⢠Microphone Input
Soft Limiting
⢠Optical Output SelectSelects between ADAT or optical S/PDIF for the MicroDock
⢠S/PDIF Output FormatSelects between S/PDIF or AES/EBU format for S/PDIF. This
The Mic/Hi-Z inputs have built-in āsoft limitersā which
automatically turn down the gain before the signal
overloads the A/D converters. The soft limiters allow you to
record a hotter signal without fear of clipping.
This control turns the soft limiters On or Off. See
the Best Possible Recording for additional information
about the soft limiters.
ADAT Output. The coaxial S/PDIF Output is disabled when
S/PDIF optical is selected.
sets the S/PDIF-AES status bit, but does not affect the signal
level.
Output Level
Settings
Optical
Output
Select
S/PDIF
Output
Format
Making
E-MU PCI Digital Audio System33
Page 34
4 - The PatchMix DSP Mixer
f Physical input strips
are shown with BLUE text.
f Host input strips are
shown with WHITE text.
f The Input Type will
turn RED if the input is not
available. (The MicroDock
may be disconnected.)
Input Mixer Strips
Input Mixer Strips
PatchMix DSP Input Mixer Strips are stereo except for the MicroDock Mic/Line inputs.
Each input mixer strip can be divided into four basic sections.
⢠Insert SectionEffects, EQ, External/Host Sends & Returns can be inserted into the signal path.
⢠Pan ControlsThese controls position the signal in the stereo sound ļ¬eld.
⢠Aux SendsUsed to send the signal to sidechain effects or to create separate mixes.
⢠Volume Control Controls the output level of the channel.
Mono/Stereo
Input Type
Insert Section
Pan Controls
Aux Sends
Channel
Volume
Control
Input Type
The very top of the strip is labeled
mono or stereo and displays the type
of the assigned input. Input mixer
strips can be added as desired and can
be conļ¬gured to input the following:
⢠Physical input = Hardware
(Analog/SPDIF/ADAT).
⢠Host Input = Software
(Direct Sound, WAV, ASIO source)
Inserts
You can drag and drop effects from the
Effects Palette or Right-click to insert a
Physical or ASIO Send or Send/Return
A Peak Meter, Trim Control or Test
Signal can also be inserted by RightĀclicking on the Insert section.
Pan Controls
These controls allow to you position
the channel in the stereo sound ļ¬eld.
Dual controls on stereo strips allow
you to position each side indepenĀdently.
Aux Sends
These controls send the signal to
sidechain effect processors such as
reverb and delay. They can also be used
to create separate mixes for the artist or
for recording.
This screen shows a mono strip on the left and a
stereo strip on the right.
34Creative Professional
Mute/Solo
Buttons
Scribble Strip
Vol ume Control
Controls the output level of the strip
into the main/monitor mix bus.
Mute/Solo Buttons
These convenient buttons allow you to
solo or mute selected channels.
Scribble Strips
Click inside the scribble strip and type
a name of up to eight characters.
Page 35
Mixer Strip Creation
f CDs & MP3s: The
WAVE 1/2 strip is used
to playback CDs,
Windows Media Player,
and Direct Sound.
PatchMix DSP is a dynamically conļ¬gurable mixer. Each mixer sessioncan contain an
arbitrary number of strips up to a limit set by the number of available input sources and
available DSP resources.
⢠Host refers to a computer application such as Cubase.
⢠Physical refers to a hardware input or output such as an output jack.
ī To Add a New Strip:
1. Click on the New Mixer Strip button. See Overview of the Mixer. The New Mixer
Strip Input Dialog appears:
4 - The PatchMix DSP Mixer
Mixer Strip Creation
f Adding or deleting a
strip ādefragmentsā the
effect/DSP resources. If
any effect you wish to
add is unavailable
(greyed-out), try deleting
an unused strip to free up
resources.
2. Select the desired input to the mixer strip from the following choices:
⢠Physical Source:Analog or digital input (Analog, ADAT, S/PDIF)
⢠Host - ASIO Source inputStreaming audio from an ASIO software application.
Physical: Dock Mic/Line24-bit monophonic analog input on the MicroDock.
Physical: Dock In24-bit stereo analog input on the MicroDock.
Physical: Dock S/PDIF 2 channel digital audio from the S/PDIF input on the
Physical: Dock ADAT2 channel (x4 strips) digital audio from the ADAT input on
HOST SOURCEFunction
Host ASIO Output Source
From software application
Host Windows Source
From Windows
3. Select Pre-Fader Aux Sends or leave the box unchecked for Post-Fader Aux Sends.
4. Click OK to create a new strip or Cancel to cancel the operation.
E-MU PCI Digital Audio System35
MicroDock.
the MicroDock.
2 channel digital audio from an ASIO source (software app).
ASIO: 1/2, 3/4, 5/6, 7/8, 9/10, 11/12, 13/14, 15/16
Direct Sound, WDM, Windows Media
(Sound generated or handled by Windows.)
WAVE 1/2 - Default stereo source such as game sound, CD
player, beep sounds, etc.
WAVE 3/4, WAVE 5/6, WAVE 7/8 - Additional WDM channels
f See āPre or Post Fader
Aux Sendsā on page 46.
Page 36
4 - The PatchMix DSP Mixer
Mixer Strip Creation
ī To Delete a Mixer Strip:
1. Click the top of the mixer strip you wish to delete. A red border appears around
the strip, indicating that it is selected.
2. Click on the Delete Mixer Strip button, or right-click and choose Delete, or use the
Delete key on the PC keyboard. See Overview of the Mixer.
Multichannel WAVE Files
The 1616 supports 2 channels of WAVE recording and 8 channels of multichannel
WAVE playback. The WAVE channels are available for the following types of WDM
devices:
⢠Classic MME
⢠DirectSound
⢠Direct WDM / Kernel Streaming (KS)
DirectSound and the WDM/KS interfaces allow up to Eight channels of Wave Out while
the classic MME interface only exposes 2 channels.
The WAVE channels operate at all sample rates. For additional information about WDM
behavior at high sample rates, see page 87.
192kHz/96kHz DVD-Audio disks are protected against digital copying. Most DVDĀAudio disks contain duplicate 48kHz audio tracks which will play back on the 1616.
Windows Media Player/DVD/Surround Sound Playback
Select DirectSound as the output format when using Windows Media Player and other
DVD player applications.
Eight channel WAVE playback supports either 5.1, 6.1 or 7.1 surround audio. However,
the 1616 is best suited to play 5.1 surround, since it only has 6 analog outputs. (You
could play back 7.1 surround audio by using an external S/PDIF to Analog Converter.
Create a 7/8 WAVE strip and insert a Send to S/PDIF Out.)
The chart below shows how to connect the outputs for 5.1 surround sound playback.
Multichannel WAVE to Surround Sound Speaker Channels
E-DSP WAVE 1/2Front Left / Front Right1L = FL 1R = FR1 (Tip = FL Ring = FR)
E-DSP WAVE 3/4Center / Subwoofer3L = C 3R = Sub3 (Tip = C Ring = Sub)
E-DSP WAVE 5/6Rear Left / Rear Right2L = RL 2R = RR2 (Tip = RL Ring = RR)
E-DSP WAVE 7/8Side Left / Side Right N/AN/A
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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 āThe Effects Paletteā.Any number of
effects can be inserted in series.
The Inserts also have the unique ability to patch into ASIO/WAVE and external
equipment. ASIO/WAVE Sends, External Sends and External Send/Returns can be
dropped into the insert section to route the signal anywhere you want.
The Insert/Patch Bay is incredibly ļ¬exible. 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. Thatās it! That input is now available in your ASIO software.
The following types of inserts can be selected.
Hardware EffectReverb, EQ, Compressor, Flanger, etc. using PatchMix DSPās effects
which do not load your CPU.
Host ASIO SendSplits off the signal and sends it to an ASIO host input such as a
software audio recorder or anything that uses ASIO.
ASIO Direct
Monitor
Ext. Send/ReturnSends signal to a selected external output, then returns it to the chain
External SendSends the signal to an external output. See āTo Add a Send Insert:ā
Peak MeterPeak meters allow you to monitor the signal level anywhere in the
Trim PotYou can insert a gain control with up to 30 dB of gain or attenuation.
Test ToneThis special insert outputs a calibrated sine wave or noise source,
Sends the signal to a selected ASIO host input, then returns a selected
ASIO host output to the chain.
via a physical input.
.
chain. See ā
A peak level meter and phase inverter are also included.
See ā
which can be used to track down audio problems.
See ā
Meter Insertsā.
Trim Pot Insertā.
Test Tone/Signal Generator Insertā.
4 - The PatchMix DSP Mixer
Mixer Strip Creation
f You have to create an
ASIO strip or ASIO Send in
order to activate these
ASIO channels in your
software.
Working with Inserts
The Inserts are one of most powerful features of the PatchMix DSP system as they allow
you to conļ¬gure the mixer for a wide variety of applications.
ī To Add an Effect to an Insert Location:
1. Press the FX button. The effects palette appears.
2. The effects are organized into categories. Click on a folder to open it.
3. Select the effect you want, drag it over the insert section, then drop it into an insert
location.
4. To rearrange the order of effects, simply drag and drop them into the desired order.
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4 - The PatchMix DSP Mixer
r
Mixer Strip Creation
The Insert Menu
Right-Clicking over the insert section brings up a pop-up selection box containing
various insert options to help you control and manage your inserts.
ī To Add a Send Insert:
This type of insert send splits the signal at the insert point and sends it out to the selected
destination. (An āASIO Sendā becomes an input on your recording application, a
āPhysical Outā goes to a pair of output jacks. the signal also continues down the strip to
the Aux Sends and main mixer outputs.)
1. Right-Click over the Insert section. A pop-up dialog box appears.
2. Select Insert Send (to ASIO/WAVE or physical output) from the list of options. The
following dialog box appears.
Input
Insert
Send
Panning
Fader
Aux 1 Bus
Aux 2 Bus
Main Output Bus
3. Choose one of the Send Outputs. Click on a destination to select it.
4. Click OK to select the output or Cancel to cancel the operation.
To ASIO, WAV o
Physical Output
ī To Add a Send/Return Insert:
This type of insert send breaks the signal at the insert point and sends it out to the
selected destination such as an external effect processor. A return source signal is also
selected which returns the signal to the channel strip after processing.
1. Right-Click over the Insert section. A pop-up dialog box appears.
2. Select āInsert Send/Return (Physical Output and Input)ā from the list of options.
The following dialog box appears.
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4 - The PatchMix DSP Mixer
t
Mixer Strip Creation
Input
Insert
Send/Return
Panning
Fader
Aux 1 Bus
Aux 2 Bus
Main Output Bus
To Physical Output
From Physical Inpu
3. Choose one of the Send Outputs. Click on a destination to select it.
4. Choose one of the Return Inputs. Click on a source to select it.
5. Click OK to select the Send and Return, or Cancel to cancel the operation.
ASIO Direct Monitor Send/Return
This type of insert send breaks the signal at the insert point and sends it out to the
selected ASIO Host Input destination (such as Cubase or Sonar). A return source signal
is also selected which returns the signal to the channel strip from an ASIO Host Output.
The ASIO Direct Monitor Send/Return is unique in that it utilizes ASIO 2.0 zero-latency
monitoring. In order to utilize this feature, Direct Monitoring must be enabled in the
audio recording application.
While recording, the Direct Monitor Send/Return routes the signal to the recording
application, but monitors directly from the input to eliminate latency. During playback,
the recording application automatically switches the Direct Monitor Send/Return to
monitor the recorded track.
ī 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.
InputInput
Recording
Software
Direct MonDirect Mon
Recording
Software
RecordingPlayback
The Direct Monitor Send/Return also allows the recording application to control
volume and pan. Normally when using direct monitor recording youāll 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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4 - The PatchMix DSP Mixer
Mixer Strip Creation
ī To Add an ASIO Direct Monitor Send/Return:
1. Right-Click over the Insert section. A pop-up dialog box appears.
2. Select Insert ASIO Direct Monitor from the list of options. The following dialog
box appears.
3. Choose one of the Send Outputs. Click on a destination to select it.
4. Choose one of the Return Inputs. Click on a source to select it.
5. Click OK to select the Send and Return, or Cancel to cancel the operation.
Meter Inserts
Keeping track of signal levels is important in any audio system, be it analog or digital.
You want to keep the signal levels running as close to maximum in order to achieve high
resolution and low noise. On the other hand, you donāt 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 āpeak holdā type. The topmost bar in the meter holds its
highest level for a second to let you see transients that would otherwise be too quick for
the eye.
The peak meters are also color-coded to indicate the signal strength. The chart below
outlines the meanings of the colors. Avoid lighting the topmost red bar, as this indicates
distortion of the signal. Click on the clip indicator to turn it off.
Meter ColorIndicates
E RedIndicates signal clipping.
E YellowGood strong signal level.
E GreenSignal is present.
One of the most obvious uses of the insert meters is to set input levels. On the analog
inputs, the analog-to-digital converter (ADC) is one of the most critical points in the
signal path. You want the input signal level to drive the 24-bit ADCs into their optimum
range without clipping. A reading of 0dB on an input meter indicates signal clipping.
Level
10203040506070
--12dB
Each bar of the meter equals 1dB. The yellow bars begin at -12dB below full scale.
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The insert meters are also useful to monitor incoming digital signals such as ADAT,
ASIO or S/PDIF to make sure the mixer is receiving a proper signal level. Theyāre also
great for troubleshooting, since you can place them virtually anywhere in the mixer.
ī To Insert a Meter
1. Right-Click on an Insert location of the mixer strip. A pop-up dialog box appears.
2. Select Insert Peak Meter. A stereo peak meter appears in the insert location.
3. Select Effect in the Main Section. The meters are now shown in high resolution in
the TV screen.
To Set the Input Levels of a Strip
1. Select the topmost Insert location on a mixer strip and insert a meter (see above).
2.
Left-click on the meter insert to see the meter in the TV screen.
3. Feed your audio signal to the input of the mixer strip. The meter should now show
the signal level.
4. Adjust the output level of the external device (synthesizer, instrument, preamp,
etc.) feeding the MicroDock. The meter should be in the yellow region most of the
time with occasional forays into the red. If the clip indicator ever comes on, reduce
the signal level.
5. Each analog input pair has its own Input Pad (-10dBV or +4dBu) which controls the
input signal range. Changing the I/O settings can add or subtract 12dB. Check these
settings if you cannot set the proper input level. See I/O Settings.
4 - The PatchMix DSP Mixer
Mixer Strip Creation
f Input too weak?
Use -10 Input setting.
Output too weak?
Use +4 Output setting
Making the Best Possible Recording
Making a good digital recording is easier than ever thanks to the high resolution 24-bit
A-D converters on your Digital Audio System. These converters are much more forgiving
than the 12-bit or 16-bit converters of the past. Even so, to get the best performance
possible, you'll need to follow a few basic guidelines.
First, whenever you input an analog signal to the Digital Audio System, make sure that
you're feeding the A-D converters with an optimum signal level. The quality of a digital
recording is directly related to the signal level you feed into the A-D converters. If the
analog input level is set too low, you lose resolutionāif 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 āEffectā button at the top of the TV screen.
The āI/O Settingsā in the Digital Audio System allow you to set the input levels to
-10dBV (consumer equipment level) or +4dBu (professional equipment level) for each
analog input. This control sets the overall input level to match your other gear, but to get
the best possible recording you need to ļ¬ne tune the level further.
In order to supply the correct input level, youāll 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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4 - The PatchMix DSP Mixer
Mixer Strip Creation
have to go back and adjust the I/O Settings. Choose ā-10ā if the input signal is too weak
and ā+4ā if the signal is too strong.
Digital audio has NO headroom past 0dBFS (FS = Full Scale) and will āhard clipā if the
signal exceeds 0dBFS. Hard clipping sounds bad and will ruin your recording. Hard
clipping occurs because at 0dBFS, all 24 bits are turned on and the A-D cannot measure
any higher level. Analog tape, unlike digital, can be driven past 0dB, although with
some degradation of the signal.
The MicroDock includes a pair of Soft Limiters on the preamp inputs, which can be
turned on or off for each channel in the I/O Settings. The soft limiters automatically
turn down the gain whenever the signal level exceeds -6dB below Full Scale. Below this
level, the limiters are completely out of the circuit. The soft limiters allow you to encode
a hotter signal without fear of hard clipping the input. This provides increased
resolution and a better recording. When recording drums, piano and vocals, occasional
peak transients can be tamed by the soft limiters, allowing you to supply the best
possible signal into the MicroDockās ultra-high-quality A-D converters.
The Digital Audio System includes Insert āTrim Potā controls, but since they adjust the
signal level AFTER the signal has been digitized, this will not recover any lost resolution.
Itās far better to set the input level correctly in the ļ¬rst 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 itās
much better to boost the signal level before the A/D converters in order to get maximum
resolution and signal-to-noise ratio from the converters.
The phase invert switch inverts the polarity of the signal. It is generally used to correct
for balanced lines and mics that are wired backwards.
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ī To Add a Trim Pot Insert
1. Right-Click over any of the Insert sections. A pop-up dialog box appears.
2. Select Insert Trim Control from the list of options. A Trim Pot insert appears in the
insert location.
3. Click on the Trim Pot insert to view and adjust the controls in the TV screen.
4. To move the Trim Pot to another location, simply drag and drop it into the desired
position.
Test Tone/Signal Generator Insert
The test tone/signal generator insert is a handy troubleshooting aid which outputs a
calibrated sine wave, white noise or pink noise. This tool, in combination with an insert
meter, allows you to accurately measure the signal gain or attenuation of an internal or
external device. The test tone can also be quite handy for tuning up musical instruĀments.
Signal Type
(Sine wave, White or Pink Noise)
4 - The PatchMix DSP Mixer
Mixer Strip Creation
f Musical Note Freq.
A = 440 Hz
B = 493.88 Hz
C = 523.25 Hz
D = 587.33 Hz
E = 659.26 Hz
F = 698.46 Hz
G = 783.99 Hz
Sine Wave Oscillator Frequency
Test Signal Output Level
The Sine Wave Oscillator frequency is variable from 20Hz-20kHz. The level is variable
from off to +30dB.
White Noise is a mixture of all frequencies in the audio spectrum at the same average
level (analogous to white light in the visible spectrum).
Pink Noise provides equal power distribution per octave. (White noise has more power
in the higher octaves.) Pink noise and white noise are useful as wideband sound
sources.
ī Using the Test Tone and Meter Inserts for Troubleshooting
Sometimes itās 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 ļ¬rst setting up your recording software.
1. Right-Click over the Insert section in question. A pop-up dialog box appears.
2. Select Insert Test Tone/Signal Generator from the list of options. A Test Tone insert
appears in the insert location.
3. Click on the Test Tone insert to view and adjust the controls in the TV screen.
4. To move the Test Tone to another location, simply drag and drop it into the
desired position.
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4 - The PatchMix DSP Mixer
Mixer Strip Creation
Managing Your Inserts
ī To Delete an Insert:
1.
Right-Click over the Insert you wish to delete. A yellow line around the insert
location indicates that it is selected. A pop-up dialog box appears.
2. Select Delete Insert to remove the selected insert or select Delete All Inserts to
remove all inserts.
3. The insert(s) are deleted from the insert chain.
ī To Bypass an Insert:
Inserts can be bypassed if you want to temporarily hear the audio without the effect or
insert. Bypass can also be used to turn off a Send Insert.
Method #1
1. Click on the Effect (in the Insert section) and select Effect in the TV display.
2. Click the Bypass button.
Method #2
1. Right-Click over the Effect you want to bypass (in the Insert section). A pop-up
dialog box appears.
2. Select Bypass Insert from the list of options.
f Tip: Select the Insert
and press the Delete key
to delete the plug-in from
the strip.
ī To Bypass All Inserts:
All Inserts in a strip can also be bypassed with a single command.
1. Right-Click over the Effect you want to bypass (in the Insert section). A pop-up
dialog box appears.
2. Select Bypass All Inserts from the list of options.
ī To Solo an Insert:
Inserts can also be soloed. Solo bypasses all the other inserts in the strip and allows you
to hear only the soloed effect. This feature is very useful when adjusting the effect
parameters.
Method #1
1. Click on the Effect (in the Insert section) and select Effect in the TV display.
2. Click the Solo button.
Method #2
1. Right-Click over the Effect you want to Solo (in the Insert section). A pop-up dialog
box appears.
2. Select Solo Insert from the list of options.
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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 donāt want the submix to be combined
into the main mix.
Aux Send and Return values can also be changed by typing directly into the displays.
4 - The PatchMix DSP Mixer
Mixer Strip Creation
Input
Pan
Fader
Mute
Amt
Amt
Sidechain Diagram
(Post-Fader Aux Sends)
Send
Amount
Aux Bus 1
Chain
Send
Amount
Aux Bus 2
Chain
Main / Monitor Bus
Return
Amount
Side
Return
Amount
Side
Output
Other Uses of the Aux Sends
You can think of the Aux Sends as two extra mixing buses because thatās exactly what
they are. These two mixes can be routed anywhere, such as to a physical output or an
ASIO pair. You could route one of the Aux buses to the Monitor out to create a monitor
mix while sending the main mix off to your audio recording software.
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4 - The PatchMix DSP Mixer
Mixer Strip Creation
Pre or Post Fader Aux Sends
When you create a New Mixer Strip you have the option to place both Aux Sends after
the channel volume fader and mute control or you can place them before the fader and
mute. Post-Fader turns down the send level as you lower the volume of the strip. With
Pre-Fader selected, you may still hear the effected signal returning from one of the Aux
Buses with the volume fader turned down.
With the Pre-Fader box selected, the Aux Send levels are completely unaffected by the
Level Fader and Mute settings. The Pre-Fader setting allows you to create two completely
different mixes using the Aux Buses since the signal levels of this mix wonāt be affected
by the fader settings.
Input
Volume Fader & Mute does NOT affect Send Levels
Pre-Fader Aux Send
Pan
Return
Amount
Return
Amount
Amt
Amt
Aux Bus 1
Send
Amount
Side
Chain
Send
Amount
Side
Fader
Aux Bus 2
Chain
ī In order to change a
strip from pre-fader to
post-fader or vice-versa,
you have to delete the
strip and create a new
one.
Pan
Fader
Mute
Mute
Input
Amt
Amt
Main / Monitor Bus
Post-Fader Aux Send
Volume Fader & Mute affects both Aux Send Levels
Send
Amount
Aux Bus 1
Send
Amount
Aux Bus 2
Output
Return
Amount
Side
Chain
Return
Amount
Side
Chain
Main / Monitor Bus
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Page 47
Level, Pan, Solo & Mute Controls
The Pan control comes before the Level Control
Pan Controls
Aux Send
Amount
Controls
Level Control
Mute & Solo
Buttons
Scribble Strip
and Aux Sends in the signal ļ¬ow. On stereo strips
we use an unconventional pan section with two
pan pots ā 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
expectāpress the button and the sound from that
channel is cut off. Pressing the Solo button while
the Mute button is pressed allows you to hear the
channel until solo is turned off.
The Solo button allows you to listen to only that
channel while muting the rest of the inputs. If
multiple solo buttons are pressed, you will hear all
soloed channels and the non-soloed channels will
all be muted.
The mute status is remembered if a muted channel
is soloed. When the channel solo is turned off, the
channel reverts to being muted.
The Level Control for the strip is an attenuation
control that can also provide up to +12dB of gain.
0db is the unity gain setting. You can also type
numeric values into the displays to set the level.
At the very bottom is the Scribble Strip text area,
into which you can type any short piece of text,
thus naming the strip, i.e. āvocalsā, ābassā,
ādrumsā and so on.
4 - The PatchMix DSP Mixer
Mixer Strip Creation
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4 - The PatchMix DSP Mixer
Main Section
Main Section
Physical/Host
Select Buttons
āTVā Screen
View
Selection
Buttons
Aux
Insert
Section
Master
Aux Send
Amounts
Main
Insert
Section
Output
Fader &
Meters
Master Aux
Return
Amounts
Sync &
Sample Rate
Indicators
Monitor Controls
Session Name
The main section contains all controls for controlling the main mix elements as well as
a āTV screenā for viewing the input/output routing or parameters of the selected insert.
The three buttons across the top of the main section select what is shown on the TV
display. Input and output routings are graphically displayed. When an insert is selected
(by clicking on the insert), the screen shows the available parameters for the currently
selected insert.
Below the TV screen is the Aux Bus section where effects, effects chains or other inserts
can be assigned to the two aux buses. Send and return levels can be individually
controlled for each of the two Aux Buses.
The Aux 1 and Aux 2 buses are fed by the two Aux Sends on each mixer strip. The Master
Send Level control on Aux bus 1 and 2 can be used to attenuate or boost the signal
going into the Auxiliary Inserts. There is also a Master Return Level to control the
amount of the effected signal that will be returned into the main mix.
The Main Bus can also have a chain of effects inserted. (You might put an EQ here to
equalize your entire mix or add an ASIO or WAVE send to record the mix.) Note that the
Main Output level control comes before the Monitor Level so that you can control the
monitor level without affecting the level of your recording mix or main mix. There is a
stereo peak meter that indicates the signal strength for the main mix.
The Monitor section has a volume, balance, and a mute control to cut off the monitor
output.
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TV Screen & Selectors
The āTV screenā at the top of the main section is a multi-function display and control
center for the input and output routings and effect controls. The three buttons at the top
of the display select the current function of the displayāEffect, 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 āNo
Insertā.
Most effects have a wet/dry mix parameter to control the ratio of effect to plain signal.
The wet/dry setting is stored with the effect preset. The parameter set varies with the type
of effect. See āList of Core Effectsā for detailed information about the individual effects.
4 - The PatchMix DSP Mixer
Main Section
E Note: Effects have to
be placed into an insert
location before you can
program them.
Effect Display
View Button
Wet/Dry Mix Control
Effect Location
Effect Bypass &
Solo Buttons
Effect Parameters
User Preset Section
When a Send or a Send/Return insert is selected with the effects display enabled, the TV
screen shows you where the Send is going and where the Return is coming from. The
bypass or solo buttons at the top of the display are available for Send/Return type inserts
only.
Send Destination
Return Source
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4 - The PatchMix DSP Mixer
Main Section
Input
Selecting the Input display view shows a graphic representation of the PatchMix DSP
Mixer inputs. This screen is only a display, unlike the Effects and Outputs screens, which
allow you to make routing changes. Input routing changes are made by adding mixer
strips. See Mixer Strip Creation.
The input routings are divided into two categories: Physical Inputs and Host Inputs.
Select either category by clicking on the Physical or Host button.
Physical Input DisplayHost Input Display
Return
Amount
f The Input and Output
displays make it much
easier to understand the
signal routings of a
complex mixer setup.
f Tip: Clicking on any
of the input routings in
the TV display highlights
the corresponding mixer
strip.
Output
Selecting the Output display view shows a graphic representation of the PatchMix DSP
Mixer outputs. The output routings are divided into two categories: Physical Outputs
and Host Outputs. Select either category by clicking on the Physical or Host button.
Physical Output DisplayHost 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 āphysicalā 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 āInsert Sectionāfor information on how to connect the inserts.
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Auxiliary Effects & Returns
The section immediately below the TV Screen is where you assign the Auxiliary Effects.
In a traditional mixing console, auxiliary effects sends are used to send part of the signal
to outboard effect devices, then return the effected signal back into the mix using the
effect returns. This is called a sidechain routing because the aux signal takes a detour
through the effects before being summed back into the main mix.
Sidechain effects are usually effects that you might want applied to several channels,
such as reverb. Effects such as EQ and compressors are usually NOT used as sidechain
effects because they can cause unpredictable results when returned to the main bus.
4 - The PatchMix DSP Mixer
Main Section
f The Wet/Dry mix
setting in the effect
should normally be set to
100% when the effect is
inserted as a sidechain
effect. This is because the
Aux Return Amount will
control the wet/dry mix.
Send
Amount
Sidechain
Effects
Return
Amount
Input
Pan
Fader
Mute
Aux
Amt
Input
Aux
Amt
Aux Bus
Sidechain Diagram
(Post-Fader Aux Sends)
Side
Return
Amount
Output
Send
Amount
Chain
Main Bus
You can also use the Auxiliary Sends as two extra mix buses. By turning the Aux Return
amount all the way down and dropping an Insert Send into the chain, you can send the
Auxiliary bus to any output you wish. See āInsert Sectionā for more information.ā
Sync/Sample Rate Indicators
The Sync/Sample rate Indicators show the current
sessionās 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 āEXTERNALā.
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
āLOCKEDā LED will extinguish.
If the external clock source makes a radical sample rate change from the lower rates of
44.1k/48k to a higher rate or between any of the higher rates, the hardware automatiĀcally switches to internal 48kHz clock until the proper external clock is restored. The
āLOCKEDā LED will be off and the two units are NOT synchronized. Always check the
āLOCKEDā LED when using an external clock source to make sure you are sampleĀlocked.
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4 - The PatchMix DSP Mixer
Main Section
Output Section
Clip Indicators
Main
Insert
Section
Main Output Level Fader
Output Level
Meters
Sync/Sample
Rate Indicators
Monitor
Mute
Monitor
Balance
Monitor
Volume
Main Inserts
The main inserts allow you to apply effects to the main stereo signal coming out of the
mixer (both mains and monitor). You might want to apply EQ or a compressor here.
These inserts work just like the other insert locationsājust 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 ampliļ¬ers or other equipment.
MAIN MIX
10
10
20
20
30
30
0dB
-12dB
Output Level Meters
This stereo bar-graph meter reļ¬ects the digital level at the output of the mixer. The
topmost red bar represents 0 dB or a full-scale digital signal. The peaks hold for a
moment so that short transients can be monitored. Each bar = 1dB.
40
40
50
50
LR
Monitor Output Level
This control adjusts the monitor output level. Keep in mind that since the monitor level
control comes after the Main Output Fader, nothing will be heard from your monitors if
the main level is turned down.
Monitor Balance Control
This control sets the relative volume of the stereo monitor outputs and works just like
the balance control on your home music system. This control is primarily used to make
the volume from each speaker sound equal if you are not sitting exactly in the center of
the two speakers.
f Tip: The volume
control on a multimedia
computer keyboard can
be used to control the
Monitor Output Level on
PatchMix.
Monitor Output Mute
This button completely cuts off the monitor output and provides a convenient way to
instantly kill all sound without having to re-adjust the monitor level later. When the
telephone rings, just hit the monitor mute to cut the noise.
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5 - Effects
Overview
PatchMix DSP comes complete with a host of great core DSP effects including
Compressors, Delays, Choruses, Flangers and Reverb. Each 32-bit effect has various
parameters for editing, as well as factory presets. You can also create and save as many of
your own effect presets as you wish.
Since the effects are implemented in hardware, they donāt 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 ļ¬nite limit to how many effects you can use at the same time. As you use up
the PatchMix DSP resources, certain effects will appear āgrayed outā and cannot be
added to the mixer. Complex effects such as reverb use more DSP resources than say a
1-Band EQ. If you continue to add effects, all of the DSP resources will eventually be
used up.
The Effects Palette
Click the FX button on the toolbar to bring up the Effects Palette. The Effects Palette
contains two types of folders. The āCore Effectsā folder contains the effect algorithms
themselves. This folder cannot be modiļ¬ed. The other folders contain āEffects Chainsā,
consisting of two or more effects grouped together. You can also add, delete, or modify
Effects Chains and the folders that contain them. For more information on Effects
Chains, see āFX Insert Chainsā on page 54.
5 - Effects
Overview
f Saving a session
ādefragmentsā the effect/
DSP resources. If you
have used all your effects
and need another, try
saving the session.
New Folder buttonImport/Export FX Button
Effect Categories
Core Effects
(Single Effect)
Multi-Effects
(Effect Combinations)
Distortion Lo-ļ¬
Drums & Percussion
Environment
Equalization
Guitar
Multi Effects
Reverb
Synths & Keys
Vocal
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5 - Effects
The Effects Palette
ī To Select an Effect
1. Click the FX button to bring up the Effects Palette. The effect palette contains
numerous folders containing effects presets. Click on any folder to open it.
2. Select the effect you wish to use by clicking on it with the left mouse button and
while continuing to hold the mouse button, drag the effect into the desired location
on the PatchMix DSP mixer screen and release the mouse button. Multi effects
contain several effects along with their parameter settings.
3.
If you want to change the order of effects, simply Left-click and drag the effect to the
desired location. Drag the effect to the area above or below the ļ¬nal destination and
release the mouse button to move the effect.
ī To Edit an Effect
1. Click on the Insert Location containing the effect you wish to edit. The effect
controls now appear on the TV screen.
2. Edit the effect parameters as desired.
ī To Delete an Effect
1. Right-click on the Insert location containing the effect you wish to delete and a pop-
up list appears.
2. Select āDelete Insert(s)ā from the top of the list. The effect will be deleted.
f The order of effects in
a chain can have a big
effect on the sound.
This icon will
appear when you drag
an effect to a new
location.
FX Insert Chains
FX Insert Chains can be used to save several effects and their settings into a single multiĀeffect. When an effects chain is selected and dropped into an insert location, all the
effects with control settings are copied as a single entity. Once dropped into an insert
location, the effects are totally separate just as if you had placed them individually.
ī To Save FX Insert Chains
1. Select two or more effects and place them into any consecutive insert locations.
2.
Set the effect parameters the way you want them, including wet/dry mix settings.
3. Right-click to bring up the list of options.
4. Select āSave FX Insert Chainā. The New FX preset dialog box appears.
5. Select a category folder where your preset will be placed, and enter a new preset
name for your FX Chain.
f Trim pots, peak meters
and test tone generators
will also be included in
the FX chain.
6. Select a folder where your new preset will be placed, then type in a new preset name
and click OK. Your preset is now saved.
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Creating, Renaming & Deleting Categories or Presets
There are several utilities to help you organize your effects presets.
ī To Create a New Preset Category
You can create your own category folders to help organize your effects presets.
1. Left-click on the New Folder icon at the top of the Effects Palette. A pop-up dialog
box appears asking you to āEnter the Name of the New Category.ā
⢠Alternatively, you can Right-click over an Effects Folder, which calls a pop-up
dialog box with the option to āCreate New Category.ā
2. Type in a name for your new folder.
3. Click OK to create a new folder or Cancel to cancel the operation.
ī To Delete an Effect Category or Preset
1. Right-click on the category folder you wish to delete. A pop-up selection box
appears.
2. Select āDelete Categoryā. A popup dialog box appears warning you that this action
will delete all presets in the folder.
3. Click OK to delete the folder or Cancel to cancel the operation.
5 - Effects
The Effects Palette
ī To Rename an Effects Category
1. Right-click on the category folder you wish to rename. A pop-up selection box
appears.
2. Select āRename Categoryā. A pop-up dialog box appears, asking you to āEnter New
Category Name.ā
3. Click OK to rename the folder or Cancel to cancel the operation.
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5 - Effects
The Effects Palette
Importing and Exporting Core FX Presets and FX Insert Chains
These utilities make it easy to import or export your FX Presets and FX Insert Chains.
You can share presets with your friends or download new presets from the Internet.
ī To Import Core FX Presets
This option imports complete folders of Core FX presets into the E-MU PatchMix DSP
folder (normally located here: āC:\Program Files\Creative Professional\E-MU PatchMix
DSP\Core Effectsā). If the name of an imported FX preset exactly matches a preset you
already have, a number will be appended to end of the imported preset name.
1. Click the Import/Export FX Library button from the FX Palette.
2. Select Import FX Library. The āBrowse for Folderā window appears.
3. Choose the folder where the Core FX presets you wish to import are located.
4. The selected folder of Core FX presets will be copied into the Core Effects folder of
PatchMix DSP.
ī To Import FX Category Folders
This option imports complete category folders of FX Chains into the E-MU PatchMix
DSP folder (normally located here: āC:\Program Files\Creative Professional\E-MU
PatchMix DSP\Effect Presetsā). If the name of an imported FX preset exactly matches a
preset you already have, a number will be appended to end of the imported preset
name.
1. Click the Import/Export FX Library button from the FX Palette.
2. Select Import FX Category. The āBrowse for Folderā window appears.
3. Choose the folder where the FX Chains you wish to import are located.
4. The selected folder of FX Chains will be copied into the Effect Presets folder of
PatchMix DSP.
ī To Export your Core FX Presets
This option exports your Core FX presets to a folder of your choice.
1. Click the Import/Export FX Library button from the FX Palette.
2. Select Export FX Library. The āBrowse for Folderā window appears.
3. Choose a destination location for the Core FX presets, then press OK.
4. The Core FX presets will be copied to the selected destination.
ī To Export your FX Category Folders
This option exports a single category of FX chains to a folder of your choice.
1. Click the Import/Export FX Library button from the FX Palette.
2. Select Export FX Category. A pop-up dialog box appears asking you to āChoose the
FX Category to be exportedā.
3. Choose the desired FX Category to export. Press OK to continue or Cancel to
cancel the operation.
4. The āBrowse for Folderā window appears. Choose a destination location for the
Core FX presets, then press OK.
5. The FX Chains will be copied to the selected destination.
56Creative Professional
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FX Edit Screen
Click on an FX Insert to display the parameters for that effect. If an insert effect is not
selected, the FX display will read āNo Insertā.
Most effects have a wet/dry mix parameter to control the ratio of effect-to-plain signal.
The wet/dry setting is stored with the FX preset. The effect parameters vary with the type
of effect. Generally if an effect is placed in an Aux Send, the wet/dry mix in the effect
should be set to 100% wet since the Aux Return amount controls how much effect is
applied.
The User Preset section is located at the bottom of the FX Edit screen. User presets are
variations of the main effect and can be edited, deleted, renamed or overwritten as you
wish.
5 - Effects
FX Edit Screen
E Note: Effects have to
be placed into an insert
location before you can
program them.
Effects Display
View Button
Wet/Dry Mix Control
ī To Bypass an Insert:
Effect Location
Effect Bypass &
Solo Buttons
Effect Parameters
User Preset Section
Inserts can be bypassed if you want to temporarily hear the audio without the effect or
insert. Bypass can also be used to turn off a Send Insert.
Method #1
1. Click on the Effect (in the Insert section)
2. Click the Bypass button in the TV display.
Method #2
1. Right-click over the Insert you want to bypass (in the Insert section). A pop-up
menu appears.
2. Select āBypass Insertā from the list of options. The insert effect name will āgray-outā
to indicate that the insert effect is bypassed.
ī To Solo an Insert:
Inserts can also be soloed. Solo bypasses all the other inserts in the strip and allows you
to hear only the soloed effect. This feature is very useful when adjusting the effect
parameters.
Method #1
1. Click on the Insert Effect (in the Insert section).
2. Click the Solo button in the TV display.
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5 - Effects
FX Edit Screen
Method #2
1. Right-click over the Insert Effect you want to Solo (in the Insert section). A pop-up
menu appears.
2. Select āSolo Insertā from the list of options. The other Insert Effect names in the
strip will āgray-outā to indicate that they are bypassed.
ī To Bypass ALL
All the inserts in a strip can be bypassed with a single command.
1. Right-click over any Effect in the Insert section. A pop-up menu appears.
2. Select āBypass All Insertsā from the list of options. All the insert names will be
āgrayed-outā to indicate that they are bypassed.
ī To Un-Bypass ALL
All the inserts in a strip can also be un-bypassed with a single command. This command
works even if only some of the effects are bypassed.
1. Right-click over any Effect in the Insert section. A pop-up menu appears.
2. Select āUn-Bypass All Insertsā from the list of options. All the insert names will light
to indicate that they are active.
User Preset Section
Each core effect has a set of User Presets, that you can use to store your favorite effect
parameter settings. Weāve 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 modiļ¬ed settings. User presets
stay with the Mixer application regardless of which Session is open.
Click here for Edit Menu
Click here to Select Presets
ī To Select a User Preset
1. Select the FX display in the TV screen.
2. Select the desired insert effect, highlighting it. The effect parameters appear in the TV
screen.
3. Click on the icon on the preset menu. A drop-down preset list appears.
4. Select a preset from the list.
ī To Create a New User Preset
1. Select the FX display in the TV screen.
2. Select the desired insert effect, highlighting it. The effect parameters appear in the TV
screen.
3. Click on the Edit button. A pop-up menu appears.
4. Select New. A pop-up dialog box appears asking you to name the new preset.
5. Name the preset and click OK. Your new preset is now saved.
E To copy or share User
Presets, you must save
them as FX Palette
effects.
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ī To Delete a User Preset
1. Select the user preset you wish to delete from the user preset menu.
2. Click on the Edit button. A pop-up menu appears.
3. Select Delete. A pop-up dialog box appears asking you to conļ¬rm your action.
4. Click OK to delete the preset or No or Cancel to cancel the operation.
ī To Rename a User Preset
1. Select the user preset you wish to rename from the user preset menu.
2. Click on the Edit button. A pop-up menu appears.
3. Select Rename. A pop-up dialog box appears asking you to rename the preset.
4. Type in the new preset name, then click OK to rename the preset or Cancel to cancel
the operation.
ī To Overwrite or Save a User Preset
This operation allows you to overwrite an existing preset with a newer version.
1. Select the user preset you wish to modify from the user preset menu and make any
changes you wish.
2. Click on the Edit button. A pop-up menu appears.
3. Select Overwrite/Save. The current preset will be overwritten with the new settings.
5 - Effects
FX Edit Screen
Core Effects and Effects Presets
The Core Effects cannot be removed or copied. Effect presets (stored in āC:\Program
Files\Creative Professional\E-MU 1616\E-MU PatchMix DSP\Effect Presetsā) can be
copied, e-mailed or shared like any other computer ļ¬le.
f Hint: You can open
the effects presets with
āNotePadā or other word
processor to view and
edit the name and
parameters.
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5 - Effects
List of Core Effects
List of Core Effects
Stereo ReverbFrequency ShifterMono Delay 750
Lite ReverbAuto-WahMono Delay 1500
CompressorVocal MorpherMono Delay 3000
Leveling Amp1-Band Para EQStereo Delay 100
Chorus1-Band Shelf EQStereo Delay 250
Flanger3-Band EQStereo Delay 500
Distortion4-Band EQStereo Delay 750
Speaker SimMono Delay 100Stereo Delay 1500
RotaryMono Delay 250
Phase ShifterMono Delay 500
DSP Resource Usage
There are two main factors which determine the total number of effects available for use
at any given time: Tank Memory and DSP Instructions. Using too much of either
resource will cause effects to be unavailable (grayed out) in the FX menu. In addition,
the strips themselves use DSP Instructions, so only create strips that you actually need.
Tank memory is the memory used by delay-based effects such as reverb and digital
delays. All the reverbs and delays aside from the Mono Delay 100 and Stereo Delay 100
use varying amounts of tank memory.
The DSP instructions are used by all the effects. Effects with multiple stages, such as
multi-band EQs or the speaker simulator use more DSP instructions than a 1-Band EQ.
Tank memory tends to get used ļ¬rst, and so weāve provided many delay line effects to
allow maximum conservation of this precious resource. Use only the longest delay you
actually need.
f Tip: Saving a session
ādefragmentsā the effect/
DSP resources. If you
have used all your effects
and need another, try
saving the session.
The chart below shows three possible effects combinations. These were created by using
up the reverb resources ļ¬rst. 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 1No.Example 2No.Example 3No.
Stereo Reverb2Lite Reverb5Stereo Reverb1
4-Band EQ43-Band EQ5Lite Reverb2
3-Band EQ21-Band EQ4Stereo Delay 15001
1-Band EQ6Compressor1Mono Delay 2501
Compressor6Mono Delay 15001Compressor6
Chorus1Mono Delay 2501Chorus2
Mono Delay 15001Auto-Wah1Flanger2
4-Band EQ3
3-Band EQ3
Total Effects 22Total Effects18Total Effects21
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Core Effects Descriptions
f
1-Band Para EQ
+15dB
Boost
Width
+
Gain
-
Cut
-15dB
Center
Frequency
ParameterDescription
GainSets the amount of cut (-) or boost (+) of the selected frequency
band. Range: -15dB to +15dB
Center FrequencySets the range of frequencies to be cut or boosted with the Gain
control. Range: 80Hz to 16kHz
BandwidthSets the width of the frequency range for the Center Frequency
band that will be cut or boosted by the Gain control.
Range: 1semitone to 36 semitones
This single band parametric equalizer is useful
when you just want to boost or cut a single range
of frequencies. For example, if you just want to
brighten up the lead vocal a bit, you might
choose this EQ. This EQ offers up to ±15dB cut
or boost.
5 - Effects
Core Effects Descriptions
1-Band Shelf EQ
This single band shelving equalizer is useful when you just want to boost or cut a single
range of frequencies at the high or low end of the spectrum. For example, if you just
want to add a little more bass, thereās 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 Shelforā¦High Shel
+15dB
Corner
Freq
Corner
Freq
+
Boost
Gain
-
Cut
-15dB
ParameterDescription
Shelf TypeAllows you to choose either low shelving or high shelving EQ.
GainSets the amount of cut (-) or boost (+) of the shelf.
Corner Frequency Sets the frequency where the signal begins getting cut or boosted
Frequency
Range: -15dB to +15dB
with the Gain control. Range: 80Hz to 16kHz
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5 - Effects
f
Core Effects Descriptions
3-Band EQ
This versatile equalizer provides two shelving ļ¬lters 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 ShelfMid BandHigh Shel
+24dB
Corner
Freq.
Corner
Freq.
E Note: The Wet/Dry
Mix control on an
equalizer should normally
be set to 100% wet or
unpredictable results may
occur.
Width
Center
Boost
Gain
Cut
+
-
-24dB
Frequency
ī Setting up a Parametric EQ
1. Turn up the gain on the band you are working with. This allows you to easily hear
the effect of the ļ¬lter.
2. Reduce the bandwidth if you are working with a mid-band.
3. Adjust the Center Frequency to āzero-inā on the frequencies you wish to boost/cut.
4. Set the Gain to a positive value to boost frequencies or to a negative value to cut out
frequencies.
5. Widen the Bandwidth to create a more natural sound.
6. Adjust and tweak as needed.
ParameterDescription
High Shelf GainSets 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 GainSets the amount of cut (-) or boost (+) of the mid frequency band.
Range: -24dB to +24dB
Mid Freq. 1Sets the range of frequencies to be cut or boosted with the Mid
Gain control. Range: 200Hz to 3kHz
Mid BandwidthSets 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 GainSets 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
62Creative Professional
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4-Band EQ
This 4-band equalizer provides two shelving ļ¬lters at the high and low ends of the
frequency range and two fully parametric bands in the center. Up to ±24 dB of boost or
cut is provided for each band.
Note: The Wet/Dry Mix control on an equalizer should normally be set to 100% wet or
unpredictable results may occur.
For more information about setting up a parametric EQ, see page 62.
5 - Effects
Core Effects Descriptions
Low-ShelfMid 1-BandHigh-Shelf
Corner
Frequency
Width
Center
Frequency
Boost
Gain
Cut
+
-
Mid 2-Band
Corner
Frequency
Width
Center
Frequency
Frequency
ParameterDescription
High Shelf GainSets 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 GainSets 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 BandwidthSets 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 GainSets 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 BandwidthSets 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 GainSets the amount of cut (-) or boost (+) of the low frequency shelf.
Range: -24dB to +24dB
Low Corner Freq.Sets the frequency where the signal begins getting cut or
boosted with the Low Gain control. Range: 50Hz to 800Hz
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5 - Effects
Core Effects Descriptions
Auto-Wah
This effect creates the sound of a guitar wah-wah pedal. The āWahā ļ¬lter 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 ļ¬lter 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 āenvelopeā, or volume contour, controls the frequency of
the bandpass ļ¬lter 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 ļ¬lter sweeps
back at a rate determined by the Release setting.
The wah direction allows the ļ¬lter to be swept either up or down in frequency. Use a
higher Center Frequency setting when the wah direction is down.
Auto-Wah Filter
Center
Frequency
Bandwidth
Envelope
Sensitivity
Input
Wave
Sweep Range
AttackRelease
Envelope Follower
ParameterDescription
Wah DirectionAllows you to sweep the wah up or down.
Env. SensitivityControls how closely the wah sweep follows the input signal.
Range: -12dB to +18dB
Env. Attack TimeSets the starting rate of the āwahā sweep.
Range: 0ms to 500ms
Env. Release TimeSets the ending or release rate of the āwahā sweep.
Range: 10ms to 1000ms
Sweep RangeControls the amount of āwahā sweep. Range: 0% to 100%
Center FrequencySets the initial bandpass ļ¬lter frequency.
Range: 80Hz to 2400Hz
BandwidthSets the width of the bandpass ļ¬lter. Range: 1Hz to 800Hz
64Creative Professional
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Chorus
t
An audio delay in the range of 15-20 milliseconds is too short to be an echo, but is
perceived by the ear as a distinctly separate sound. If we now vary the delay time in this
range, an effect called chorus is created, which gives the illusion of multiple sound
sources. A slight amount of feedback serves to increase the effect. A very slow LFO rate is
usually best for a realistic effect, but a faster LFO rate can also be useful with minimal
LFO depth (.2). Since this is a stereo chorus, an LFO phase parameter is included which
can be used to widen the stereo image.
ParameterDescription
DelaySets the length of the delay. Range: 0ms to 20ms.
FeedbackSets the amount of delayed signal that will be recirculated through
the delay line. Range: 0% to 100%
LFO RateSets the frequency of the low frequency oscillator.
Range: .01Hz to 10Hz
LFO DepthSets how much the LFO affects the delay time. Increases the
animation and amount of the chorus effect. Range: 0% to 100%
LFO WaveformSelectable between Sine or Triangle wave.
LFO L/R PhaseControls the stereo width by adjusting the phase difference of the LFO
waveform between left and right channels. Range: -180° to +180°
5 - Effects
Core Effects Descriptions
Compressor
In its simplest form, an audio compressor is just an automatic gain control. When the
volume gets too loud, the compressor automatically turns it down. Compressors are
useful in musical applications because they allow you to record a āhotterā signal
without overloading the recording device.
Since the compressor turns down the gain of the signal, you might wonder how can it
make the signal level stronger. A Post Gain control allows you to boost the output gain
of the compressor in order to make up for the gain reduction. The overall level is higher
and only turned down when the signal level gets too loud. This level is called the
Threshold, which just happens to be the most important control on the compressor.
Signal path = Stereo
In
Delay
Level
VCA
Ou
Control
Threshold
Ratio
E-MU PCI Digital Audio System65
Attack
Release
Post Gain
Page 66
5 - Effects
Core Effects Descriptions
Basic Controls
The three main controls of a compressor are the Ratio control, the Threshold control and
the Gain control.
If the signal falls below the Threshold, no processing will take place. Signals exceeding
the Threshold will have gain reduction applied as set by the ratio control. This
important control allows you to dial in the range of amplitudes you want to tame. For
example, if youāre 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 ampliļ¬es the signal after it has been compressed to bring it back
up in volume. If you donāt 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 āpumpingā effect as the compressor turns on and off. Donāt
make the release time too long, however, or the compressor wonāt 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 ālook into the futureā up to 4 milli-
seconds in order to anticipate upcoming peaks in the signal. This is accomplished
of course, by inserting delay into the signal path. This lookahead technique
allows the use of slower attack times without missing signal peaks. This
parameter is especially effective on drums and percussion.
The Input Meter allows you to monitor the strength of your input signal. Always try to
boost the signal before the compressor if you can.
The Compression Meter shows the amount of gain reduction being applied. Since this
meter displays how much the gain is being turned down, the meter moves from right to
left, instead of left to right like a normal meter.
ParameterDescription
ThresholdThreshold 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
RatioSets the ratio of input signal level to output signal level, or
āhow muchā compression will be applied. Range: 1:1 to ā:1
Post GainAmpliļ¬es the signal after it has been compressed to bring up the
volume. Range -60dB to +60dB
66Creative Professional
Page 67
ParameterDescription
t
Attack TimeControls how quickly the gain is turned down after the signal
exceeds the threshold. Range .1ms to 500ms
Release TimeControls how fast the gain is returned to its normal setting after
the signal has fallen below the threshold.
Range: 50ms to 3000ms
Pre-DelayAllows the use of slower attack times without missing signal peaks.
Range: 0ms to 3 ms
Input MeterAllows you to monitor the strength of the input signal.
Gain Reduction MeterShows 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 āfuzz boxā style,
clipping distortion which is particularly effective on guitar, bass, organs, electric pianos
or whatever.
The input signal ļ¬rst passes through a lowpass ļ¬lter. 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 āhow
muchā distortion will be added. A bandpass ļ¬lter follows the distortion generator. The
EQ Center control lets you select a particular band of frequencies to be output. The EQ
Bandwidth controls the width of the center frequency band. Finally, a gain control
allows you to make up for any gain loss through the effect.
Use the Wet/Dry mix control in conjunction with the Edge control to reduce the
amount of distortion, or go wild and turn everything to 11!
5 - Effects
Core Effects Descriptions
Lowpass
Filter
In
Signal path = Stereo
Distortion
LP Filter
Cutoff
ParameterDescription
Pre EQ LP CutoffControls the amount of high frequency audio admitted to
the distortion. Range: 80Hz to 24kHz
EdgeSets the amount of distortion and new harmonics
generated. Range: 0-100
GainSets the output volume of the effect. Range: -60dB to 0dB
Post EQ Center Freq.Sets the frequency of the output bandpass ļ¬lter.
Range: 80Hz to 24kHz
Post EQ BandwidthSets the width of the output bandpass ļ¬lter.
Range: 80Hz to 24kHz
Edge
Bandpass
Filter
Ou
GainEQ BW
EQ Center
E-MU PCI Digital Audio System67
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5 - Effects
t
Core Effects Descriptions
Flanger
A ļ¬anger is a very short delay line whose output is mixed back together with the original
sound. Mixing the original and delayed signals results in multiple frequency cancellaĀtions known as a comb ļ¬lter. Since the ļ¬anger is a type of ļ¬lter, 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 ļ¬lter and deepens the effect.
Feedback
In
Flanger
Signal path = Stereo
Delay
ParameterDescription
DelaySets the initial delay of the ļ¬anger in .01 millisecond increments.
This parameter allows you to ātuneā the ļ¬anger to a speciļ¬c
frequency range. Range: .01ms to 4ms
FeedbackControls how much signal is recirculated through the delay line
and increases resonance. Negative values can produce intense
ļ¬anging with some signals. Range 0% to 100%
LFO Rate Sets the speed of the ļ¬anger sweep. Range: .01 Hz to 10Hz
LFO DepthSets how much the LFO affects the delay time. Increases the
animation and amount of the ļ¬anging effect. Range 05 to 100%
LFO WaveformSelectable between Sine or Triangle wave.
LFO L/R PhaseControls the stereo width by adjusting the phase difference
between the left and right sweeps. Range: -180° to +180°
LFO
Waveform
Phase
Ou
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Page 69
Freq Shifter
This unusual effect is sometimes called āspectrum shiftingā or āsingle sideband
modulationā. Frequency shifting shifts every frequency in the signal by a ļ¬xed 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 āmusicalā harmonies.
This isnāt to say that the frequency shifter canāt 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 speciļ¬ed amount from .1 Hz to 24
kHz. You can also shift pitch up on one side and down on the other if you wish.
Comparison between Pitch and Frequency Shifting
5 - Effects
Core Effects Descriptions
f You can also type in
exact frequencies to a
resolution of 1/10 Hz.
Harmonic
Original
(Hz)
Pitch Shifted
(100 Hz)
Frequency Shifted
(100 Hz)
1200 300300
2400600500
3600900700
48001200900
5100015001100
6120018001300
7140021001500
8160024001700
ParameterDescription
FrequencySets the number of Hz that will be added or subtracted with every
harmonic in the signal. Range: .01Hz to 24kHz
Left DirectionSets pitch shift up or down for the left channel.
Right DirectionSets pitch shift up or down for the right channel.
E-MU PCI Digital Audio System69
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5 - Effects
Core Effects Descriptions
Leveling Amp
The ļ¬rst compressors developed in the 1950ās 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 ālookahead delayā to apply gentle gain
reduction. Because of this delay, the leveling amp is not suitable for applications which
require realtime monitoring of the signal. This smooth and gentle compressor is
designed to be used in situations where delay does not pose a problem, such as
mastering a mix or compressing prerecorded stereo material.
Post Gain is the only control on the leveling amp. This control is used to make up the
volume lost by the compression. The Compression Ratio is ļ¬xed 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 GainAmpliļ¬es the signal after it has been compressed to
bring up the volume. Range 0dB to 36dB
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Lite Reverb
Reverberation is a simulation of a natural space such as a room or hall. The Lite Reverb
algorithm is designed to simulate various rooms and reverberation plates while using
fewer DSP resources than the Stereo Reverb. Up to ļ¬ve Lite Reverbs can be used at once.
Decay time deļ¬nes the time it takes for the reļ¬ected sound from the room to decay or
die away. The diagram below shows a generalized reverberation envelope.
5 - Effects
Core Effects Descriptions
Early Reflections
After a short pre-delay period, the echoes from the closest walls or ceiling are heard.
These ļ¬rst echoes, or Early Reļ¬ections, vary greatly depending on the type of room. Some
time after the early reļ¬ection cluster ends, the actual Reverberation (a dense cloud of
complex wall reļ¬ections) 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 reļ¬ections and the reverberation cloud.
High frequency energy tends to fade away ļ¬rst 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 reļ¬ective and have less high frequency damping. Rooms ļ¬lled 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 āboominessā of the room.
ParameterDescription
Decay TimeSets the reverb decay time. Range: 0% to 100%
HF Decay FactorSets the rate at which high frequencies die away. The high
LF Decay FactorSets the rate at which low frequencies die away. The low
Early Reļ¬ectionsSets the volume of the initial wall reļ¬ections.
ReverberanceSets the amount of scattering of the early reļ¬ections and
Reverberation
frequencies last longer as the percentage is increased.
Range: 0% to 100%
frequencies last longer as the percentage is increased.
Range: 0% to 100%
Range: 0% to 100%
the reverberation cloud. Range: 0% to 100%
Time
E-MU PCI Digital Audio System71
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5 - Effects
t
Core Effects Descriptions
Mono Delays - 100, 250, 500, 750, 1500, 3000
A delay line makes a copy of the incoming audio, holds it in memory, then plays it back
after a predetermined time. The delay number refers to the maximum delay time that
can be produced by the delay line. The six lengths, from 100 ms to 3 seconds, allow you
to make the most efļ¬cient 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 ļ¬anging and comb ļ¬lter 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 ļ¬ltering produced by the delay line. Comb ļ¬
A High Frequency Rolloff ļ¬lter in the feedback path cuts some of the high frequency
energy each time the audio goes through the delay line. This simulates the natural
absorption of high frequencies in a room and can also be used to simulate tape-based
echo units.
The Wet/Dry mix controls how loud the echoes are in relation to the original signal.
ltering: See page 68.
Feedback
HF
Rolloff
L In
L Out
Delay
R In
Delay Time
ParameterDescription
Delay Time
Mono Delay 100
Mono Delay 250
Mono Delay 500
Mono Delay 750
Mono Delay 1500
Mono Delay 3000
FeedbackSets the amount of delayed signal that will be recirculated through
High Freq. RolloffDamps high frequencies in the feedback path.
Sets the length of the delay in milliseconds.
(.01ms. minimum increment between settings)
Range: 1 millisecond to 100 milliseconds
Range: 1 millisecond to 250 milliseconds
Range: 1 millisecond to 500 milliseconds
Range: 1 millisecond to 750 milliseconds
Range: 1 millisecond to 1.5 seconds
Range: 1 millisecond to 3 seconds
the delay line. Range: 0% to 100%
Range: 0% to 100%
R Ou
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Page 73
Phase Shifter
t
A phase shifter produces a ļ¬xed 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 1970ās 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 ļ¬xed multi-notch ļ¬lter is
created.
Feedback
In
Phase
Signal path = Stereo
Shifter
Ou
5 - Effects
Core Effects Descriptions
LFO Center
LFO
LFO Rate
ParameterDescription
LFO CenterSets the initial offset of the LFO and changes the position of the
peaks and notches. Range: 0% to 100%
FeedbackIncreases the depth of the notches and height of the peaks.
Range: 0% to 100%
LFO RateControls the sweep rate of the Low Frequency Oscillator.
Range: .01Hz to 10Hz
LFO DepthControls how much the Center Frequency is swept by the LFO.
Range: 0% to 100%
WaveformSelects a Sine or Triangle wave for the LFO
LFO L/R PhaseControls 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.
ParameterDescription
SpeedSwitches between slow or fast rotor speeds with
acceleration and deceleration as the speed changes.
E-MU PCI Digital Audio System73
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5 - Effects
Core Effects Descriptions
Speaker Simulator
The Speaker Simulator provides realistic guitar speaker responses and is designed for use
with guitar, bass or synthesizer. Twelve popular guitar amp speaker cabinets are
modeled.
There is only one parameter on this effect. Just select the speaker you want and listen.
Normally this effect should be used with the Mix control set to 100%.
Speaker TypeDescription
British Stack 1 & 2Modeled from a British 8-speaker high power ampliļ¬er stack.
British Combo 1-3Modeled from a British 2-speaker combo ampliļ¬er.
Tweed Combo 1-3Modeled from an American, 1950ās era, 2-speaker combo ampliļ¬er.
2 x 12 ComboModeled from an American, 1960ās era, 2-speaker combo ampliļ¬er.
4 x 12 ComboModeled from an American, 1960ās era, 4-speaker ampliļ¬er set.
Metal Stack 1 & 2Modeled from a modern era, power ampliļ¬er stack.
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Stereo Delays - 100, 250, 500, 750, 1500
t
The Stereo Delays are true stereo delay lines in that the left and right channels are kept
entirely separate from each other. The delay number refers to the maximum delay time
that can be produced by the delay lines. The ļ¬ve different lengths, from 100 ms to 1.5
seconds, allow you to make the most efļ¬cient use of the effect memory resource.
Because the left and right channels can have different delay times, you can create a
panning effect by setting one delay long and the other short. Very short delay times
combined with a high feedback amount can be used to create monotone roboticĀsounding effects. Using the longer stereo delays, you can āoverdubā musical lines one
on top of the other with the feedback control turned up.
Feedback
HF
Rolloff
In
5 - Effects
Core Effects Descriptions
Delay
Signal path = Stereo
L Delay
Time
ParameterDescription
Left Delay TimeSets the length of the delay for the left channel in milliseconds.
Right Delay TimeSets the length of the delay for the right channel in milliseconds.
FeedbackSets the amount of delayed signal that will be recirculated through
High Freq. RolloffDamps high frequencies in the feedback path. Range: 0% to 100%
(.01ms. minimum increment between settings)
Range: 1 millisecond to 100 milliseconds
Range: 1 millisecond to 250 milliseconds
Range: 1 millisecond to 500 milliseconds
Range: 1 millisecond to 750 milliseconds
Range: 1 millisecond to 1.5 seconds
the delay line. Range: 0% to 100%
R Delay
Time
Ou
E-MU PCI Digital Audio System75
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5 - Effects
Core Effects Descriptions
Stereo Reverb
Reverberation is a simulation of a natural space such as a room or hall. The stereo reverb
algorithm is designed to simulate various halls, rooms and reverberation plates.
Decay time deļ¬nes the time it takes for the reļ¬ected sound from the room to decay or
die away. The diagram below shows a generalized reverberation envelope.
Time
Early Reflections
Late Reverb
After a short pre-delay period, the echoes from the closest walls or ceiling are heard.
These ļ¬rst echoes, or early reļ¬ections, vary greatly depending on the type of room. Some
time after the early reļ¬ection cluster ends (late reverb delay), the late reverberation (a
dense cloud of complex wall reļ¬ections) 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 ļ¬rst 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 reļ¬ective and have less high frequency damping. Rooms
ļ¬lled 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 āboominessā of the room.
ParameterDescription
Decay TimeSets the length of the Late Reverb. Range 1.5 to 30 seconds
Early Reļ¬ections LevelSets the volume of the initial wall reļ¬ections.
Range: 0% to 100%
Early/Late Reverb BalAdjusts the balance between early refections and late reverb.
Range: 0% to 100%
Late Reverb DelaySets the time between early reļ¬ections and the onset of the late
reverb cloud. Range: 1ms to 350ms
DiffusionSets the amount of scattering of the late reverb cloud.
Range: 0% to 100%
High Freq. DampingSets the rate at which high frequencies die away.
Range: -10.0 to +3.0 damping factor
Low Freq. DampingSets the rate at which low frequencies die away.
Range: -10.0 to +3.0 damping factor
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Vocal Morpher
This unique effect allows you to select two vocal phonemes and morph between them
using an LFO. Phonemes are the consonants and vowels we use in articulating speech
sounds and these sounds are very distinctive and evocative. 30 different phonemes are
available and these can be shifted up or down in pitch for even more effects.
To use the Vocal Morpher, you just select Phoneme A and Phoneme B from the list of
thirty. Now the LFO automatically morphs back and forth between the two selected
phonemes, creating interesting vocal articulations. The rate of the LFO is adjustable and
you can select between Sine, Triangle or Sawtooth waveforms. The sine and triangle
waves fade smoothly. The sawtooth wave gradually fades, then jumps abruptly back.
When the frequency of the A or B Phonemes is shifted up or down, entirely new effects
can be produced. These frequency controls can also be used to tune the phoneme
frequencies to the range of audio you are processing.
Phoneme B
5 - Effects
Core Effects Descriptions
Frequency
Time
Phoneme A
List of Available Phonemes
AE IOUAA
AEAHAOEHERIH
IYUHUWBDF
GJKLMN
PRSTVZ
ParameterDescription
Phoneme ASelect 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 BSelect 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 RateControls how fast the phonemes morph back and forth.
Range: .01Hz to 10Hz
LFO WaveformSelects the waveform for the morph: Sinusoid, Triangle, Sawtooth
E-MU PCI Digital Audio System77
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5 - Effects
E-MU PowerFX
E-MU PowerFX
The hardware-accelerated effects of the E-MU Digital Audio System can also be used as
VST inserts in Cubase. E-MU PowerFX allow you to use PatchMix DSP effects from
within Cubase with minimal load on your CPU.
E-MU PowerFX incorporate smart time alignment technology which automatically
compensates for system latencies and ensures proper synchronization of audio
throughout the VST chain (if the host application supports this feature).
E-MU PowerFX On/OffPreferences
Input Signal Present
FX Parameters
ī E-MU PowerFX are not
available at 88.2kHz,
96kHz, 176.4kHz and
192kHz sample rates.
f Cubase SX/SL 2.0,
Nuendo and Sonar (using
the Cakewalk VST
adapter 4.4.1) implement
VST 2.X auto delay
compensation.
FX Palette
FX Inserts
Output Signal Present
FX Presets
Preset Editing
ParameterDescription
PowerFX On/OffEnables or bypasses E-MU PowerFX.
FX PaletteSelect from a single āCoreā effect or a Multi -Effect.
FX InsertsDrop Effects from the FX Palette here.
Signal Present LEDsThese indicators turn blue to show the presence of input and
output signals.
FX ParametersSelect the desired effect in the center insert section, then adjust
the wet/dry mix and parameters for the effect.
FX PresetsSelect from the list of preprogrammed effect presets here.
Preset EditingClick here to Save, Delete, Rename or Overwrite a User Preset.
See the āUser Preset Sectionā for more information
78Creative Professional
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ParameterDescription
PreferencesThe Preferences menu allows you to:
⢠Toggle the Tooltips On or Off
⢠Extra Buffers - Check this box if excessive stuttering occurs when
using E-MU PowerFX in your VST Host application. This box
should be checked when using Fruity Loops.
⢠Render Mode - Induces realtime rendering in applications
which do not support realtime rendering (WaveLab, SoundĀForge).
5 - Effects
E-MU PowerFX
ī To Setup & Use E-MU PowerFX:
Setup Cubase or Cubasis
1. Launch Cubase or Cubasis.
2. Instantiate E-MU PowerFX in an Insert or Aux Send location within Cubase.
3. Press the Insert Edit button in Cubase to bring up the E-MU PowerFX plug-in
window shown on the previous page.
Setup E-MU PowerFX
4. Make sure the blue button is illuminated, indicating that E-MU Power FX is on.
The blue āSignal Presentā indicators will be illuminated if E-MU PowerFX is
properly patched into a signal path.
5. Drag the desired effects from the Effects Palette to the center Insert strip.
6. Click on the Effect you wish to edit in the center Insert Strip (it will highlight in
yellow), then adjust the effects parameters in the right section of the window.
7. You can also select or edit User Presets from the section below the FX parameters.
See the ā
Add Delay Compensation
If you are using Cubase VST 5.1, you will have to insert an E-Delay Compensator into
any other audio tracks to keep them time-aligned.
8. Simply insert an E-Delay Compensator plug-in into the same insert location you
used for E-MU PowerFX on any other audio tracks. Thatās it.
User Preset Sectionā for more information.
(if needed)
ī Using any driver other
than āE-MU ASIOā may
produce undesirable
results when using E-MU
PowerFX.
E-MU PCI Digital Audio System79
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5 - Effects
E-MU PowerFX
Automating E-MU PowerFX
E-MU PowerFX can be automated in Cubase LE (or other recording host) just like any
other VST effect. When āWrite Automationā is activated in Cubase, control changes
made in the PowerFX window during playback will be recorded on a special
āAutomation Subtrackā. When āAutomation Readā is activated, the recorded control
changes will be played back.
ī To Record E-MU PowerFX parameter changes in Cubase LE
1. Add E-MU PowerFX as a Channel Insert.
2. Rewind the song and enable āAutomation Writeā by pressing the WRITE button
on, illuminating it. (Refers to Cubase LE. If you are using another application,
refer to the documentation.)
3. Bring the E-MU PowerFX window to the front and select the Effect you want to
automate. The effect parameters appear in the TV screen. Make sure the blue āOnā
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 ļ¬nished.
6. Disable āAutomation Writeā and enable āAutomation Readā . Playback the song
to hear and view your changes.
7.
To edit Automation, ļ¬rst enable both āAutomation Writeā and āAutomation Readā
and press Play. Cubase LE begins overwriting as soon as you change a control.
8. If you donāt like the results and want to try again, select Show Used Automation
from the Project menu. The Automation Subtrack appears. Next, click in the
Parameter Display and select Remove Parameter.
Note: This only erases one automation parameter from the Automation Subtrack.
To erase multiple control edits, repeat the procedure above. See the Cubase LE
manual for more speciļ¬c information about automation editing.
ī Steinberg Cubasis
does not have the control
automation feature.
ī Once you have
recorded or drawn
automation, do not
delete or move effects
from the Insert Strip.
Doing so will result in
unpredictable behavior.
E-MU PowerFX Resource Availability
Because different collections of VST plug-ins and PatchMix Sessions can be run simultaĀneously, it is possible to load a Cubase Song or PatchMix Session for which resources
are not available. If DSP resources are NOT available for an existing setup:
⢠E-MU PowerFX loads a Hardware I/O Path and simply passes audio through
without any effects. The effects insert slot(s) in E-MU PowerFX will be āredded outā.
⢠If no Hardware I/O Paths are available, the plug-in will be disabled and run in a soft
pass-through mode. The effects insert slot(s) in E-MU PowerFX will be āgrayed outā.
⢠If DSP resources ARE available, but no Hardware I/O Paths are available, the plug-in
will run in soft pass-through mode.
⢠If the sample rate is changed in the middle of a E-MU PowerFX session, E-MU
PowerFX plug-ins will be bypassed, since the hardware effects cannot operate at
88kHz, 96kHz, 176.4kHz or 192kHz.
80Creative Professional
Page 81
E-MU PowerFX Compatibility Chart
5 - Effects
E-MU PowerFX
Application NameCompatible?NoteRender
Extra
Buffers
Steinberg Cubase VST 5.1YesOffOff
Steinberg Cubase SX 1YesOffOff
Steinberg Cubase SX 2YesInstrument
OffOff
Freeze triggers
error if
not in render
mode.
Steinberg Cubase LEYesOffOff
Steinberg Cubase SLYesOffOff
Steinberg WaveLab 4YesOnOff
Steinberg WaveLab Lite (ver 4)Ye sOnOff
Steinberg WaveLab 5NoPops & clicks
OnEither
may occur.
(Try 8 buffers at
1024)
Sony Acid 4YesOnOff
Sony Vegas 5YesOnOff
Sony SoundForge 7NoPower FX
OnOff
crashes when
launched.
Adobe Audition 1.5NoAudio
AnyAny
distortion &
immediate
lockup.
FruityLoops Studio 4.5YesOffOn
Ableton Live 3.5NoDistortion
OnOff
when FX
parameters are
changed.
Cakewalk Sonar 3YesOffOff
E-MU PCI Digital Audio System81
Page 82
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 ļ¬le 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 āASIO Buffer Latencyā setting located in
the device Setup dialog box. Depending on your setup, you may have to
increase or decrease the Buffer Latency settings to ļ¬nd the setting that works.
⢠Instead of rendering with E-MU PowerFX, bounce the E-MU PowerFX processed
tracks to another track in realtime.
⢠Check āRealtime Renderā in the Render dialog box when using Cubase LE,
Cubase SX2 or Cubase SL2. This setting will give the best results.
Tips for using Freeze Mode on Cubase LE
ā¢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 āFreezingā 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 ļ¬rst few audio
buffers as they are fed by WaveLab to E-MU PowerFX. The problem can be eliminated by
following these guidelines.
⢠Check āRender Modeā box in the E-MU PowerFX preferences. See page 79.
ā¢We recommend that you only use the MME/WAVE E-DSP Wave [xxxx] drivers.
⢠Reduce the āBuffer Sizeā in the WaveLab, Audio Preferences dialog box. This
moves the stuttering to beginning of the ļ¬le.
⢠Pad the beginning (and/or end) of your audio ļ¬le with silence (.5 to several
seconds depending on the ļ¬le). This action causes the buffer discontinuities to
occur
before
the song begins.
82Creative Professional
Page 83
E-MU VST E-Wire
E-Wire is a special VST/ASIO Bridge which allows you to route digital audio via ASIO to
PatchMix and back again.
E-Wire VST incorporates smart time alignment technology that automatically compenĀsates for system latencies and ensures proper synchronization of audio throughout the
VST chain. In addition, E-Wire also allows you to insert outboard audio gear into the
VST environment.
5 - Effects
E-MU VST E-Wire
E-Wire has three main components:
ā¢A VST plug-in which handles the audio routing to PatchMix DSP.
⢠An ASIO mixer strip in PatchMix DSP conļ¬gured to route audio to the E-Wire
plug-in. You simply drop the effects you want to use into this strip.
ā¢For hosts that donāt support automatic delay compensation, a manual delay-
compensation plug-in can be inserted in Cubase tracks or channels that donāt use
E-Wire to compensate for ASIO delay.
The diagram below may give you a better idea of how E-Wire works:
E-Wire VST plug-in
Send to Strip
E Note: Itās easier to use
E-MU PowerFX instead of
E-Wire if you just want to
use the hardware effects.
(E-Wire was the precursor
to E-MU PowerFX.)
However, E-Wire can be
very useful because it
allows you to route VST
inserts or Sends to
Physical Inputs and
Outputs via PatchMix DSP.
Stereo Reverb
Return to VST
ASIO Send
PatchMix DSP
Strip configured
for E-Wire
E-Wire bridges the gap between hardware I/O and the VST world. The E-Wire VST plug-in sends
audio to a strip containing the desired effect. An ASIO Send routes the audio back to E-Wire VST.
E-MU PCI Digital Audio System83
Page 84
5 - Effects
E-MU VST E-Wire
ī To Setup and use E-Wire:
Setup PatchMix DSP
1. Open PatchMix DSP application.
2. Insert an ASIO Input mixer strip into PatchMix DSP. (Alternately, you can select
āNew Sessionā, select āE-Wire Exampleā and skip to step 6.)
3. Mute the strip or turn the Fader all the way down.
4. Insert an ASIO Send plug-in into one of the inserts on your ASIO strip.
5. Name your ASIO strip as an E-Wire strip.
6. Insert the desired PatchMix DSP effects into slots above the ASIO Send.
7. Save the Session.
Setup Cubase
8. Launch Cubase.
9. Instantiate E-Wire VST in an Insert or Aux Send location within Cubase.
10. Edit the E-Wire plug-in and activate the plug-in by pressing the blue button.
11. Set the ASIO Send and Return on the E-Wire plug-in to match the strip you set up
for E-Wire.
12. Done.
E-Delay Compensation
An E-Delay Compensator must be inserted into any other audio tracks that are not using
E-Wire in order to keep them time-aligned.
13. Simply insert an E-Delay Compensator plug-in into the same insert location you
used for E-Wire on any other audio tracks. Thatās it.
E-Delay Compensator
As audio is transferred back and forth between the VST host application and the E-MU
sound hardware, a delay in the audio stream is incurred. Normally this delay is compenĀsated for automatically by the host application, but not all VST host applications
support this automatic compensation.
A host will support PowerFX and E-Wireās plug-in delay compensation if it supports the
SetInitialDelay feature of the VST 2.0 speciļ¬cation.
84Creative Professional
Page 85
Currently automatic delay compensation is supported by the Steinberg 2.0 family
(Nuendo 2.x, Cubase SX 2.0), Magix Samplitude 7.x, and Sonar (using the Cakewalk
VST adapter 4.4.1), but not, unfortunately, by Steinberg Cubase VST 5.1, Cubase LE and
Cubasis.
The E-Delay Compensator utility plug-in is used to manually compensate for the
transfer delay for hosts that DO NOT support plug-in delay compensation.
The E-Delay Compensator plug-in is used to delay the ādryā tracks (tracks without a
PowerFX or E-Wire as an insert effect) or auxiliary (send) channels. For each dry track or
send, add an E-Delay Compensator plug-in to re-align the track. The E-Delay CompenĀsator is automatic and requires no user interaction to operate.
For example, consider a Cubase VST session with two audio tracks. If PowerFX or E-Wire
is applied as an insert effect to the ļ¬rst audio track, but not to the second, the ļ¬rst track
will be delayed in relation to the second track. The E-Delay Compensator should be
added as an insert effect on the second track in order to provide delay compensation.
Cubase VST or Cubasis
5 - Effects
E-MU VST E-Wire
Track 1
Insert
E-Wire
PatchMix
DSP
Track 2
Insert
Track 3
Insert
E-DelayE-Delay
E-Delay Compensator Use
For host applications that donāt 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 doesnāt 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.
E-MU PCI Digital Audio System85
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5 - Effects
E-MU VST E-Wire
In practical use, however, youāll 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 ājust in caseā you need it.
Hereās one more example of how to use the E-Delay Compensator with different
numbers of PowerFX/E-Wire sends on each track. The delay compensation on each track
must equal the track with the maximum number of PowerFX/E-Wire sends. See the
diagram below.
Cubase VST or Cubasis
Track 1
Insert
PowerFX
or E-Wire
Insert
PowerFX
or E-Wire
Track 2
Insert
PowerFX
or E-Wire
Insert
E-Delay
1
Track 3
Insert
E-Delay
2
PatchMix
DSP
Since track 1 uses two PowerFX/E-Wire inserts, the delay of all the other tracks must
equal two. Track 2 has one PowerFX/E-Wire insert and so adding one unit of E-Delay
keeps it time aligned. Track 3 doesnāt 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.
86Creative Professional
Page 87
6 - Appendix
Using High Sample Rates
Overview
When operating at 176.4k or 192k sample rates, the mixer functionality and number of
I/O channels are slightly reduced. The number of ADAT channels also decreases at the
88k/96k and 176/192k sample rates (due to the bandwidth limitations of the optical
components).
When using 88.2kHz, 96kHz, 176.4kHz or 196kHz sample rates:
6 - Appendix
Using High Sample Rates
⢠Effect processors are disabled.
⢠ADAT is reduced to 4 channels at 88k/96k, and 2 channels at
176k/192kHz.
⢠ASIO channels are reduced to 8 ASIO (4 stereo) channels at 88k/96k,
and 4 ASIO (2 stereo) channels at 176k/192kHz.
⢠At the 176.4kHz & 192kHz sample rates, the number of physical
inputs and outputs is reduced.
⢠At the 176.4kHz & 192kHz sample rates, S/PDIF optical is disabled
The ADAT optical interface was originally designed to carry 8 channels at a 48kHz
sample rate. We use the 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.
ī Selecting a 176/192k Session
The three possible input conļ¬gurations are selected by choosing a session template
containing the desired I/O from the New Session window. Once you have selected one
of the three session types, you will not be able to change to another type without
starting a new session.
(Output sends & returns are still available.)
Select the Type of Session you need
Analog & S/PDIF Session
Analog & ADAT Session
S/PDIF & ADAT Session
1. Select New Session from the PatchMix
DSP toolbar.
2. Choose the 176k/192ktab.
3. Select the Template that meets your
requirements and click OK.
E-MU PCI Digital Audio Systems87
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6 - Appendix
Using High Sample Rates
&
&
or...
or...
&
At the 192kHz sample rate, you may choose one of these three options:
1. Keep all Analog I/O, but lose S/PDIF 3. Keep S/PDIF & ADAT, but lose
2. Keep all Analog I/O, but lose ADAT Line Inputs 2L/2R & Line Outputs 3L/3R
E-MU 1616 Hardware Inputs & Outputs at 176.4k or 192k
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 ļ¬le 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 conļ¬guration, WDM audio will be played
or recorded ābit accurateā without sample rate conversion or bit truncation.
When running PatchMix at 44kHz/48kHz, if there is a mismatch between the WDM
playback audio content and the PatchMix sample rate, sample rate conversion is
performed, so that WDM audio will always be heard or recorded. Also, such non-nativeĀsample-rate audio is truncated to 16-bits.
When running PatchMix at the higher sample rates of 88.2kHz, 96kHz, 176.4kHz or
192kHz, WDM record or playback audio content must be running at the same sample
rate as PatchMix. If the sample rates are mismatched, NO AUDIO will be produced or
recorded. In other words, the WDM driver does not perform sample rate conversion of
any kind when PatchMix is running at 88.2kHz, 96kHz, 176.4kHz or 192kHz.
6 - Appendix
Using High Sample Rates
E-MU PCI Digital Audio Systems89
Page 90
6 - Appendix
Sync Daughter Card Supplement
Sync Daughter Card Supplement
SMPTE Conversion
One of the main functions of the Sync Daughter Card is to convert SMPTE (LTC) to
MIDI Time Code (MTC) and vice-versa. The term āHost MTCā refers to MTC, which is
generated or used by the host application (Cubase, etc.). MTC is also available at the
MIDI jack on the back of the Sync Card.
SMPTE Features
⢠Conversion of SMPTE to MTC quarter-frame messages & full-frame messages.
Constant quarter-frame messages are generated with steady SMPTE data input.
Occasional MIDI full-frame messages are generated when SMPTE contains data dropouts.
⢠Conversion of MTC (quarter frame & full frame) messages from the host
computer to SMPTE out.
Simultaneous SMPTE and MTC output when receiving MTC from the host computer
⢠Outputs SMPTE and MTC striping data.
SMPTE Start Time and type can be set via System Settings dialog box.
ī Warning: SMPTE and
MTC do not provide
sample sync for digital
I/O. You must use Word
Clock, S/PDIF or ADAT
sync.
SMPTE Options
When the Sync Daughter Card is installed in your system a SMPTE button in the
PatchMix DSP mixer becomes visible. Pressing the SMPTE button brings up the SMPTE
window.
Current Time
SMPTE Status/
Error Indicators
Start Striping
SMPTE Frame
Rate Settings
SMPTE
Start Time
Flywheel Mode
⢠Off ⢠Continuous
⢠Fixed ā¢1-time Jam
Stop
Striping
Output
Source
Flywheel
Amount
Word Clock
Termination
Output
Level
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6 - Appendix
Sync Daughter Card Supplement
Mode (fps)
SMPTE Striping
Stripe Button
Stop Button
Flywheel Mode
Output Level
FLY/JAM Frames
Word Clock
Termination
SMPTE/MTC
Output Source
Sets the transmitted frame rate when striping SMPTE.
Edit this ļ¬eld to set the start time in hours:minutes:seconds:frames for
striping SMPTE.
Initiates SMPTE Time Code generation at the SMPTE output beginning at
the time set in the striping display.
Stops SMPTE striping. This button also stops SMPTE when One-Time Jam
Sync has been initiated.
Selects one of the four Flywheel modes. See the descriptions below.
Sets the SMPTE output level from -10dBV (consumer) to +4dBu (pro).
If ļ¬ywheel mode is on and a dropout is detected this is the number of ļ¬yĀwheel frames that will be output before the sync card stops and chases.
Turns word clock termination on or off. Except in special cases, this conĀtrol should normally be left on. See ā
This control selects the source of the SMPTE output jack. The choices are:
Host MTC or the SMPTE Input jack (to regenerate SMPTE).
Word Clock In/Outā.
SMPTE Modes of Operation
Host Mode
The host computer is the source of synchronization. MTC messages are sent to the Sync
Daughter Card from the computer application and converted into SMPTE. MTC is also
output from the MIDI port on the Sync Daughter Card.
External Mode
SMPTE messages from SMPTE In are converted to MTC (quarter-frame messages) and
sent to the host application. This happens automatically whenever LTC is received at the
SMPTE input jack. Clean SMPTE data is also transmitted from SMPTE Out if āSMPTE
(Regenerate)ā is set.
Flywheel Mode
If the incoming SMPTE data is corrupted or missing frames, MTC code will continue to
be output if āFlywheel modeā is enabled. The ļ¬ywheel modes are described below.
Flywheel Modes
Off
Fixed 0-127
Continuous
1-Time Jam Sync
Upon any dropout, MTC stops and the Sync card monitors the input for
valid code. If valid code is again received, it chases and relocks.
Upon any dropout, MTC continues outputting Quarter-frame messages
at the same rate (ļ¬ywheeling). When a dropout is detected, this is the
number of frames that will be output before the Sync card stops outputĀting MTC and monitors the input for valid code. If valid code is again
received, it chases and relocks.
Upon any dropout, MTC continues outputting Quarter-frame messages
at the same rate (ļ¬ywheeling). The Sync card monitors the input for valid
code and continues ļ¬ywheeling until valid code is received, then relocks.
Upon any dropout, MTC continues outputting Quarter-frame messages
at the same rate (ļ¬ywheeling) without monitoring the SMPTE input until
the Stop button is pressed.
E-MU PCI Digital Audio Systems91
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6 - Appendix
ī When the MicroDock
is connected to the 1010
PCI card, the digital I/O
on the PCI card is
disabled. Use the digital
I/O on the MicroDock.
A
Sync Daughter Card Supplement
Stripe Mode
This mode is used to record SMPTE time code onto an audio track of another recorder.
SMPTE is output when the Start button is pressed in the System Settings menu and
begins at the time set by the Start Time setting. MTC is also simultaneously output from
the Sync Daughter Card MIDI out. SMPTE and MTC will continue to be output until the
Stop button is pressed.
Example SMPTE Connection
In the diagram below, Cubase is controlling the entire system by sending MTC to the
Sync Card which converts MTC to SMPTE. SMPTE is fed to the ADAT/BRC to convey the
absolute time information (hours-minutes-seconds-frames). ADAT/BRC is the word
clock master, controlling the Digital Audio System either through the embedded clock
in the ADAT optical stream or using word clock.
Cubase
1010
CARD
EDI
DAT
In
SYNC
CARD
optional
if ADAT
sync isn't
used
LTC
MTC
BRC
1
2
33
34
65
66
97
98
9
10
41
42
73
74
105
106
17
18
49
50
81
82
113
114
25
26
57
58
89
90
121
122
TRACK 1-32 TRACK 33-64 TRACK 65-96 TRACK 97-128
AUTO INPUT ALL SAFE ALL INPUT ALL CLEAR
GROUP 1 GROUP 2 GROUP 3 GROUP 4 SET GROUP
MASTER REMOTE CONTROL
7
39
71
103
15
47
79
111
23
55
87
119
31
63
95
127
8
40
72
104
16
48
80
112
24
56
88
120
32
64
96
128
PITCH UP
PITCH DOWN
PITCH MODE
MIDI UTIL
COPY SONGDELETE SONG
SET LOCATELOCATE SONG
PITCH CONTROLTAPE LOCATION
CENTS
HOURS MINUTESSECONDS FRAMES
PITCH MODE
FIXED VARIABLE
SMPTE IN MIDI IN
NAME
TEMPO MAP
RECORD XFADE
7
STUV8WXYZ0(CHARS)
SAVE SETUP
TO TAPE
4
JKL5MNO6PQR
LOAD SETUP
1
FROM TAPE
ABC2DEF3GHI
LOCATE 0
SET LOCATE
LOCATE
BARS BEATS SUB BEATS
CURSOR
RECORD
INPUT
3
4
5
6
35
36
37
38
67
68
69
70
99
100
101
102
RECORD
INPUT
11
12
13
14
43
44
45
46
75
76
77
78
107
108
109
110
RECORD
INPUT
19
20
21
22
51
52
53
54
83
84
85
86
115
116
117
118
RECORD
INPUT
27
28
29
30
59
60
61
62
91
92
93
94
123
124
125
126
NORMAL
ABSOLUTE
RELATIVEE
DROP FRAME
29.97 FPS
SMPTE
BARS
30 FPS
25 FPS
24 FPS
COPY TAPE
LOCATION
AUTO-PUNCH
DIGITAL I/O
AUTO PLAY
DISPLAY TYPEEJECT
DISPLAY MODE RESET 0 FORMAT TAPE
SMPTE START
SMPTE
PRE-ROLL POST-ROLL
LOOP TAPE OFFSETTRACK DELAY
GEN SYNC EDIT
REHEARSE
RECORDPLAYSTOPFAST FORWARDREWIND
EXT SYNC
OFFSET
PatchMix DSP set to
The Sync Card should not be used as both the SMPTE and word clock master. Word
Clock is generated by the Digital Audio System and NOT by the software application
(Cubase). SMPTE is not locked to Word Clock inside the Sync Cardāthey are
completely independent.
92Creative Professional
ADAT Sync
ADAT Optical
carries embedded
word clock
ADAT
ADAT 9-pin
REWIND FAST FWD STOPPLAY RECORD
EJECT
Page 93
MIDI Time Code (MTC)
MIDI time code is basically SMPTE time code adapted to the world of MIDI. MTC
speciļ¬es āabsoluteā location information in hours:minutes:seconds:frames, just like
SMPTE. There are two main kinds of messages in MTC: Full-frame messages and
Quarter-frame messages.
Full-frame messages are ten bytes long and are sent when SMPTE start, stops, or
relocates. Full-frame messages contain the entire SMPTE number of, hours, minutes,
seconds, frames, as well as the SMPTE type: 24fps, 25fps, 30fps non-drop, 30fps drop.
Quarter-frame messages are sent at each quarter of a SMPTE frame and only carry 1/8th
of the SMPTE time message. Quarter-frame messages require two entire SMPTE frames
to send the complete time stamp (h:m:s:f). Timing accuracy is maintained as long as the
quarter-frame messages continue to come in at a constant rate.
ī To Enable MTC:
MIDI Time code disables the use of MIDI port 2 on the MicroDock
1. Open Session Settings from the toolbar.
2. Select the MIDI tab and choose Sync Card/MTC from the MIDI options.
3. Click OK to close the window.
Since it is important to have a stable timing reference for your song or sequence, we
have given MTC its own MIDI output port on the Sync Daughter Card. This ensures that
the timing information will not be affected by other MIDI data on the line.
6 - Appendix
MIDI Time Code (MTC)
ī MTC and SMPTE do
NOT synchronize at the
sample rate and are not
locked to word clock in
any way.
SMPTE and MTC are used
to synchronize music but
do not have the required
resolution to sample-lock
digital audio.
Word Clock In/Out
Word clock provides a standardized means of synchronizing multiple digital audio
devices so that data can be transferred digitally. In order to digitally transfer from one device to another, the two devices MUST be synchronized. Clicks and pops in the
audio will result when transferring digital audio which is not synchronized.
The E-MU 1010 PCI card can be externally clocked from the ADAT input, S/PDIF input
or from the Sync Daughter card (if installed). In a digital studio, all digital devices in the system should run off the same master Word Clock.
ī To Synchronize PatchMix DSP to an External Clock Source:
1. Make sure an external clock source is connected to the E-MU Digital Audio System
hardware via the word clock, ADAT or S/PDIF input.
2. Open the Session Settings dialog box.
3. Under the System tab, select External Source, then select either word clock, ADAT
or S/PDIF.
4. Press OK to close the dialog box.
5. Check the Sync section of PatchMix DSP to verify that the Locked indicator is
illuminated.
Devices can be connected in daisy chain fashion (word clock out connected to the next
unitās word clock in) or in parallel for one or two devices, but professional digital
studios normally use a master word clock generator or āHouse Syncā with a distribution
system so that each device receives a phase-coherent and jitter-free word clock.
f Word clock, ADAT and
S/PDIF synchronize at the
sample rate and are used
to transfer digital data
between machines.
E-MU PCI Digital Audio Systems93
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6 - Appendix
A
Word Clock In/Out
Digital
Device 1
Digital
Device 3
House Sync
Generator
Digital
Device 2
A master word clock generator is preferable for larger digital setups.
Word Clock In: Receives word clock (sample clock) from another digital device such as
a digital video deck, digital recorder or digital mixer.
Word Clock Out: Sends word clock (sample clock) to another digital recorder. Word
clock is always output, whether it is generated by the internal clock or passed through
from the word clock input.
75⦠On/Off: Termination for the word clock input can be switched on or off in the
Sync Card menu of the PatchMix DSP application. Normally word clock termination
should be left on. If you have problems with a weak word clock signal, try turning termiĀnation off. See Word Clock Termination.
The diagram below shows the proper way to connect and terminate a serial word clock
chain. Using a BNC āTā connector ensures that word clock is precisely in phase for both
devices. The middle device has termination turned Off and the last device in the word
clock chain has termination turned On.
Digital
Device 4
Digital Mixer
ī When the MicroDock
is connected to the 1010
Word Clock
DAT Optical
or AES Digital
ADAT Optical
Word Clock Termination OFF
T - connector
IN
Word Clock
PCI card, the digital I/O
on the PCI card is
disabled. Use the digital
I/O on the MicroDock.
AES
E-MU 1010 CARD
EDI
IN
SYNC CARD
Word Clock Termination ON
This diagram shows the proper way to connect word clock if you donāt have a multi-output
word clock generator. The last device in a Word Clock chain should have Termination ON.
94Creative Professional
Page 95
Getting in Sync
ī When the MicroDock
is connected to the 1010
PCI card, the digital I/O
on the PCI card is
disabled. Use the digital
I/O on the MicroDock.
Whenever you connect external digital audio devices together, you need to be aware of
how they are synchronized to each other. Simply connecting digital out to digital in
doesnāt guarantee that two digital devices are synced, even if audio is being passed.
Unless you have set one to be the Master and the other a Slave, they are probably NOT
synchronized and the quality of your audio will suffer.
S/PDIF and ADAT are two commonly used digital audio formats. Both these digital
formats carry an embedded word clock which can be used to synchronize the digital
equipment. You must enable āExternal Clockā on the slave device to have clock sync!
The diagrams below show two ways to synchronize an external A/D - D/A converter to
the E-MU Digital Audio System using the ADAT lightpipe connection.
In the ļ¬rst example, only the A/D converters on the external device are being used. Only
one lightpipe is needed as long as PatchMix is set to receive its word clock signal from
the external device. The external A/D is the Master and the E-MU DAS is the Slave.
6 - Appendix
Getting in Sync
EXTERNAL
Slave
EXTERNAL
External Device supplies Master Clock
The lightpipe carries eight
channels of audio data and
an embedded clock.
12345678
Master
Set PatchMix DSP to receive:
External ADAT Sync
PatchMix DSP supplies Master Clock
This lightpipe carries eight
channels of audio data.
(via ADAT)
ADAT Out
External A/D - D/A Converter
(via ADAT)
Set External Device to receive:
ADAT Out
External ADAT Sync
Master
In the second example a second lightpipe is used to supply āembedded word clockā, as
well as eight channels of audio to the external A/D - D/A. The external device MUST be
set to receive external clock via ADAT or the units will not be synchronized. The E-MU
Digital Audio System is the Master and the external A/D - D/A is the Slave.
E-MU PCI Digital Audio Systems95
Slave
12345678
ADAT In
This lightpipe carries an
embedded clock signal
& eight channels of audio.
External A/D - D/A Converter
Page 96
6 - Appendix
Useful Information
Useful Information
Cables - balanced or unbalanced?
All inputs and outputs on the E-MU Digital Audio System are designed to use either
balanced or unbalanced cables. Balanced signals provide an additional +6dB of gain
on the inputs and are recommended for best audio performance, although unbalanced
cables are ļ¬ne for most applications. If youāre having problems with hum and noise or
just want the best possible performance, use balanced cables.
Balanced Cables
Balanced cables are used in professional studios because they cancel out noise and
interference. Connector plugs used on balanced cables are XLR (3-prong mic connector)
or TRS (Tip, Ring, Sleeve) 1/4" phone plugs.
Balanced XLR
Connectors
21
3
12
3
1 = Ground/shield
2 =
Hot (+)
3 =
Cold (-)
OutputInput
Sleeve = Ground
Balanced 1/4ā
TRS Connectors
Sleeve = Ground
Unbalanced 1/4ā
Connectors
Tip = Hot (+)
Ring = Cold (-)
Tip = Signal
ī WARNING: Do NOT
use balanced audio
cables when connecting
balanced outputs to
unbalanced inputs.
Doing so can increase
noise level and introduce
hum. Use balanced
(3-conductor) cables
ONLY if you are
connecting balanced
inputs to balanced
outputs.
Balanced cables have one ground (shield) connection and two signal-carrying
conductors of equal potential but opposite polarity. There is one āhotā or positive lead,
and a ācoldā 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 āhotā 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.
96Creative Professional
Page 97
Adapter Cables
r
1/8ā Mini-phone to 1/4ā Adapters
To connect headphones with an 1/8ā (mini-phone) plug to the headphone jack on the
MicroDock, you need a 1/8ā to 1/4ā adapter. These handy devices are available at
electronic department stores everywhere.
1/8" to 1/4"
Headphones with
1/8" plug
Cinch (RCA) to 1/4ā Adapters
Equipment (such as consumer audio gear) which uses Cinch/RCA type connectors can
be connected to the MicroDock using readily available adapter cables. These adapters
can be found at most stores that sell audio equipment.
Headphone Adapte
6 - Appendix
Useful Information
Tip = Hot (+)
Cinch/RCA
Plug
Sleeve = Ground
1/4" Phone Plug
Shaft = Ground
Tip = Hot (+)
Digital Cables
Donāt cheap out! Use high quality optical ļ¬ber Toslink (ADAT) cables. Itās 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 ļ¬ber light pipes).
Use low-capacitance, video-grade cable for coaxial S/PDIF to avoid data corruption.
AES/EBU to S/PDIF Cable Adapter
This simple adapter cable allows you to receive AES/EBU digital audio via the S/PDIF
input on the E-MU 1010 PCI card. This cable may also work to connect S/PDIF out from
the 1010 PCI card to the AES/EBU input of other digital equipment.
From AES/EBU
Device
To S/PDIF
In
N.C.
12
+
3
-
E-MU PCI Digital Audio Systems97
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6 - Appendix
V
Useful Information
Grounding
In order to obtain best results and lowest noise levels, make sure that your computer
and any external audio devices are grounded to the same reference. This usually means
that you should be using grounded AC cables on both systems and make sure that both
systems are connected to the same grounded outlet. Failure to observe this common
practice can result in a ground loop. 60 cycle hum in the audio signal is almost always
caused by a ground loop.
Phantom Power
Phantom power is a dc voltage (+48 volts) which is normally used to power the preamĀpliļ¬er 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 āridesā on top of the constant 48
volts DC. Coupling capacitors at the input of the MicroDock block the +48 volt DC
component before the signal is converted into digital form. The audio mutes for a
second when phantom power is turned on.
After turning phantom power off, wait two full minutes before recording to allow the
DC bias to drain from the coupling capacitors since the bias could affect the audio
headroom.
Balanced dynamic microphones are not affected by phantom
1
(grd)
3
Since ribbon microphones are fairly specialized and generally expensive, youāll know if
you own one. Most microphones are either of dynamic or condenser type and these are
not harmed by phantom power.
2
+48
power. An unbalanced dynamic microphone may not work
properly, but will probably not be damaged if phantom power
is left on.
Ribbon microphones should NOT be used with phantom
power on. Doing so can seriously damage the ribbon element.
Appearance Settings in Windows
Adjusting the āPerformance Optionsā in Windows will improve the screen appearance
when moving the mixer around on the screen.
ī To Improve the Appearance Settings:
1. Open the Windows Control Panel. (Start, Settings, Control Panel).
2. Select System. Select the Advanced Settings tab.
3. Under Visual Effects, select Adjust for Best Performance. Click OK.
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Technical Speciļ¬cations
Speciļ¬cations: 1616m PCI System
GENERAL
6 - Appendix
Technical Speciļ¬cations
Sample Rates
Bit Depth
Hardware DSP
Converters & OpAmps
WDM Drivers
MicroDockm Power Use
ANALOG LINE INPUTS
Type
Level (software selectable)
Frequency Response
THD + N
SNR
Dynamic Range
Channel Crosstalk
Common-mode Rejection
Input Impedance
44.1 kHz. 48 kHz, 96 kHz, 192 kHz from internal crystal
Accepts externally supplied clock from S/PDIF or ADAT
16 or 24-bits
100MIPs custom audio DSP.
Zero-latency direct hardware monitoring with effects