EXTERNAL

Connects to Audio Dock via EDI Cable S/PDIF In/Out
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. 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 congured in either Professional or Consumer mode in the Session Settings menu. The 1010 PCI card can also send and receive AES/EBU digital audio through the use of a cable adapter. See AES/EBU to S/PDIF Cable Adapter for details. The S/PDIF input and outputs are usable at the 44.1kHz, 48kHz 88.2kHz and 96kHz sample rates, but are disabled for 176.4kHz and 192kHz. The word clock contained in the input data stream can be used as a word clock source. See System Settings.
ADAT or S/PDIF Optical In/Out Firewire
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 ber light pipes for connections longer than 1.5 meters. 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 information.
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.
3 - PCI Card & Interfaces The 0202 Daughter Card
Sample Rate 44kHz/48kHz 88.2kHz/96kHz 176.4kHz/192kHz
Number of Audio Channels 8 channels of 24-bit audio 4 channels of 24-bit audio, using S/MUX standard 2 channels of 24-bit audio, using S/MUX standard

26 Creative Professional

In Word Clock Out

In SMPTE Out

MTC Out
4 - The PatchMix DSP Mixer PatchMix DSP
4 - The PatchMix DSP Mixer

PatchMix DSP

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

Overview of the Mixer

Physical Input Strips Add New Strip ASIO Input Strip Toolbar
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

TV Screen
Pan Controls Aux Sends Aux Effects Section

Volume Fader

Sync/ Sample Rate Indicators
Solo/Mute Buttons Monitor Volume/Balance /Mute Controls WAVE Strip
Controls Windows Source Audio (Direct Sound, Windows Media, etc.)
User Denable Scribble Strip

Main Inserts

Current Session Name
Main Mix Output Volume & Meters 27
4 - The PatchMix DSP Mixer Overview of the Mixer

Mixer Window

The Mixer consists of four main sections.
Application Toolbar Lets you manage sessions and show/hide the various views. Main Section Controls all the main levels, aux buses, and their inserts. This section also has a TV which shows parameters for the currently selected effect and the input/output patchbay. It also shows the sessions current sample rate and whether its set to internal or external clock. This section is located to the left of the Main Section and shows all the currently instantiated mixer strips. Mixer strips can represent Physical analog/digital inputs, or Host inputs such as ASIO or Direct Sound. Mixer strips can be added or deleted as necessary. This section can be resized by dragging the left edge of the frame. This popup window is invoked by pressing the FX button in the toolbar. Iconic representations of all effects presets are shown here, organized by category. From this window, you can drag and drop effect presets into the insert slots available on the mixer strips and main section aux buses and main inserts.

Mixer Strips

Effects Palette
A simplied diagram of the mixer is shown below.

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, MIDI, or I/O options.
The System Settings include the following:
Internal/External Clock Sample Rate Selects between internal or external word clock source as the master clock source for the system Selects the sample rate when using internal clock. Your choices are: 44.1kHz, 48kHz, 88.2 kHz, 96kHz, 176.4kHz, 192kHz. Select from: ADAT, S/PDIF or Word Clock (Sync card , only) as an external sample clock source.
E Note: if set to External without an external clock present, PatchMix DSP defaults to the internal 48kHz clock rate.
External Clock Source (ext. clock only)

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 (44.1k, 48k, 88.2k, 96k, 176.4k or 192 k) and can be distributed using a dedicated cable (word clock) or embedded into a data stream such as S/PDIF or ADAT. Common symptoms of unsynced digital audio include, random clicks or pops in the audio or failure of the digital stream to be recognized. Always check for the presence of the LOCKED indicator whenever you are using a digital interface. If an External Clock is interrupted or switched after the Session has been created (except between 44.1k <-> 48k), the LOCKED indicator will be extinguished and PatchMix will attempt to receive the external data. The two units are NOT sample locked however, and you should correct this condition to avoid intermittent clicks in the audio.
E Note: The maximum supported sample rate for S/PDIF is 96kHz.

MIDI Settings

This option allows you to use either the MIDI In jack on the rear of the AudioDock or the Sync Card MIDI jack as a MTC Output. (The MTC Out only transmits MTC.)
Dock MIDI 2 In Sync Card Enabled Selects the rear MIDI Input on the AudioDock as MIDI 2. Selects the Sync Card functions. This selection disables MIDI 2 In on the rear panel of the AudioDock. MIDI 2 Out on the rear panel of the AudioDock duplicates MIDI 1 Out in this mode.
E The SMPTE Input will not function and the Sync Card control panel will not be updated when Dock MIDI 2 In is selected.
Word Clock and SMPTE Out will operate with Dock MIDI 2 selected. Creative Professional

I/O Settings

You can set the level (-10dBV or +4 dBu) for each pair of analog outputs and the input gain setting for each pair of analog inputs. An output setting of +4 provides the most output and is compatible with professional audio gear. Balanced output cables provide a +6dB hotter signal than unbalanced cables when used with balanced inputs. Do NOT use balanced cables unless your other gear has balanced inputs. See Cables - balanced or unbalanced? in the Appendix for more information.

Comparison of -10dBV & +4dBu Signal Levels Consumer

(unbalanced)

Professional

(balanced)

+20 dBu

Clipping -->

Headroom
+ 6 dBV = +8 dBu + 2 dBV = +4 dBu -10 dBV = -8 dBu

<-- Clipping

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 rst effect location in the strip. Adjust your external equipment outputs for the optimum signal level. See To Set the Input Levels of a Strip for details.
f Input too weak? Use -10 Input setting.
Output too weak? Use +4 Output setting

Input Level Settings

Output Level Settings
E-MU 1010 Optical Input Select
E-MU 1010 Optical Output Select S/PDIF Output Format Select

Inputs +4 or -10

Selects between Consumer level (-10dBV) or Professional level (+4dBu) inputs. (Use the -10dBV setting if your input is too weak.) Selects between Consumer level (-10dBV) or Professional level (+4dBu) outputs. (The +4 dBu setting outputs a hotter level.) Selects between ADAT or optical S/PDIF for the 1010 PCI card ADAT Input. The coaxial S/PDIF input is disabled when S/PDIF optical is selected. Selects between ADAT or optical S/PDIF for the 1010 PCI card ADAT Output. The coaxial S/PDIF Output is disabled when S/PDIF optical is selected. Selects between S/PDIF or AES/EBU format for S/PDIF. This sets the S/PDIF-AES status bit, but does not affect the signal level.

Outputs +4 or -10

PCI Card Optical Input

PCI Card Optical Output

S/PDIF Optical Format
4 - The PatchMix DSP Mixer Input Mixer Strips

Input Mixer Strips

PatchMix DSP Input Mixer Strips are stereo except for the AudioDock Mic/Line inputs and the 0202 card inputs. Each input mixer strip can be divided into four basic sections.
Insert Section Pan Controls Aux Sends Effects, EQ, External/Host Sends & Returns can be inserted into the signal path. These controls position the signal in the stereo sound eld. Used to send the signal to sidechain effects or to create separate mixes.
Volume Control Controls the output level of the channel.

Mono/Stereo

Input Type
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 congured to input the following:

4 - The PatchMix DSP Mixer Main Section

Main Section

Physical/Host Select Buttons View Selection Buttons

Aux Insert Section

Master Aux Send Amounts
Master Aux Return Amounts Sync & Sample Rate Indicators

Main Insert Section

Monitor Controls Output Fader & Meters

Session Name

The main section contains all controls for controlling the main mix elements as well as a TV screen for viewing the parameters of the current 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.
E-MU Digital Audio System 49
TV Screen & Selectors
The TV screen at the top of the main section is a multi-function display and control center for the input and output routings and effect controls. The three buttons at the top of the display select the current function of the displayEffect, Inputs or Outputs.

Effect

Select the Effect display view in the main section, then click on an Effect Insert to display the effect parameters. If an insert effect is not selected, the display will read No Insert. Most effects have a wet/dry mix parameter to control the ratio of effect to plain signal. The wet/dry setting is stored with the effect preset. The parameter set varies with the type of effect. See List of Core Effects for detailed information about the individual effects.
Effect Display View Button
E Note: Effects have to be placed into an insert location before you can program them.
Effect Location Effect Bypass & Solo Buttons

Wet/Dry Mix Control

Effect Parameters

User Preset Section

When a Send or a Send/Return insert is selected with the effects display enabled, the TV screen shows you where the Send is going and where the Return is coming from. The bypass or solo buttons at the top of the display are available for Send/Return type inserts only.

Send Destination

Return Source
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.
f The Input and Output displays make it much easier to understand the signal routings of a complex mixer setup.

Physical Input Display

Host Input Display
f Tip: Clicking on any of the input routings in the TV display highlights the corresponding mixer strip.
Selecting the Output display view shows a graphic representation of the PatchMix DSP Mixer outputs. The output routings are divided into two categories: Physical Outputs and Host Outputs. Select either category by clicking on the Physical or Host button.
Physical Output Display Host Output Display
The Host Output display shows all the Insert Routings in addition to the Main Mix and Monitor out routings. Click on the desired row to make or break a physical output connection.
The Physical Output screen displays and allows you to connect the Main and Monitor outputs of the mixer to physical analog or digital outputs. Click on the box in the mix or monitor area to make (or break) a connection. The Host Output screen displays and allows you to view the Host (ASIO or WAVE) outputs of the mixer. See Insert Section for information on how to connect the inserts.

Mid 1 Gain Mid 1 Center Freq. Mid 1 Bandwidth
Low Shelf Gain Low Corner Freq.

Auto-Wah

This effect creates the sound of a guitar wah-wah pedal. The Wah 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 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 envelope, or volume contour, controls the frequency of the bandpass 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 lter sweeps back at a rate determined by the Release setting. The wah direction allows the 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

Bandwidth
Envelope Sensitivity Input Wave

Attack Release

Sweep Range

Envelope Follower

Parameter Wah Direction Env. Sensitivity Env. Attack Time Env. Release Time Sweep Range Center Frequency Bandwidth
Description Allows you to sweep the wah up or down. Controls how closely the wah sweep follows the input signal. Range: -12dB to +18dB Sets the starting rate of the wah sweep. Range: 0ms to 500ms Sets the ending or release rate of the wah sweep. Range: 10ms to 1000ms Controls the amount of wah sweep. Range: 0% to 100% Sets the initial bandpass lter frequency. Range: 80Hz to 2400Hz Sets the width of the bandpass lter. Range: 1Hz to 800Hz

Chorus

An audio delay in the range of 15-20 milliseconds is too short to be an echo, but is perceived by the ear as a distinctly separate sound. If we now vary the delay time in this range, an effect called chorus is created, which gives the illusion of multiple sound sources. A slight amount of feedback serves to increase the effect. A very slow LFO rate is usually best for a realistic effect, but a faster LFO rate can also be useful with minimal LFO depth (.2). Since this is a stereo chorus, an LFO phase parameter is included which can be used to widen the stereo image.

However, if the Highpass Freq parameter is greater than the Lowpass Frequency parameter, the passband effectively disappears, since the part of the spectrum which is above the highpass and below the lowpass is non-existent. As a result, you'll hear a rapidly attenuating bandpass response as the corner frequencies diverge. Note that while the Highpass -> Lowpass combination appears the same as the Band Pass lter, this mode is different in several important respects: The rolloff points are independently adjustable as individual frequencies rather than specied as a combination of center frequency and bandwidth. The rolloff slope of each High and Low lter can be specied separately while the Bandpass and Band Cut lters use the same slope. The Resonance of each High and Low lter can be specied separately while the Bandpass lter uses the same Resonance at high and low corner frequencies.

Highpass || Lowpass

In this mode, the Lowpass and Highpass lters are connected either in parallel, and both sets of Lowpass and Highpass parameters are exposed and independently editable to create the overall lter response. The maximum rolloff slope of each lter is limited to 24dB/octave in this mode. In Highpass || Lowpass mode, the effect does not place any limitations on the Freq parameters of one lter relative to the other. In normal use, the Highpass Freq parameter will be higher than the Lowpass Freq parameter, creating a bandcut-type response:
However, when the Highpass Freq parameter is lower than the Lowpass Freq parameter, the combined lter response is basically at, since the passbands of each lter combine to admit the entire spectrum. An exception occurs when there is resonance added to the lters - you'll hear the resonant peaks as increased gain above the otherwise at spectral response. Note that while the Highpass || Lowpass combination appears the same as the Band Cut lter, this mode is different in several important respects: The rolloff points are independently adjustable as individual frequencies rather than specied as a combination of center frequency and bandwidth. The rolloff slope of each High and Low lter can be specied separately while each side of the Band Cut lter uses the same slope. The Resonance of each High and Low lter can be specied separately while the Band Cut lter uses the same Resonance at high and low corner frequencies.

Band Pass

In this mode, the Lowpass and Highpass lters are connected in series to form a bandpass lter, whose Center Freq and Bandwidth parameters are used to generate the rolloff frequencies for the underlying Lowpass and Highpass lters. In this mode the rolloff slope on the high and low sides of the passband is symmetrical and is limited to a maximum of 24dB/octave. The Resonance is also common to both lter sections.

Creating a Ducker

Ducker
Background Signal Pan -90 (L)

Gain Cell

L Out R

Sidechain

Stereo Strip
Foreground Signal Pan +90 (R)

Example Settings

Here we have provided a few examples to show the varied uses of this useful tool. Bear in mind that these examples are simply starting points and that you will undoubtedly need to ne tune the parameters to t the program material and to suit your own taste. Increase Drum Punch: Adjust the Threshold control to control the amount of compression. Threshold: Adjust so that all hits are being compressed. Ratio: 4:1 Attack: 8 msec (Increase the time to hear more stick sound.) Release: 60 msec (Adjust according to the tempo of song.) Gain: Adjust to make up for lost volume. Soft Knee: Adjust as desired. Comp. Lookahead: This can be used instead of the Attack control. Max. Compression: Unlimited
Smoothing out the Bass Guitar Level: This setup evens out the volume and prevents the bass guitar from wandering in and out of the mix. Threshold: -24dB (adjust according to the sound) Ratio: 4:1 Attack: 8 msec Release: 70 msec Gain: +4dB (adjust according to the sound) Soft Knee: Threshold -8dB Gate: Off Comp. Lookahead: 0 msec Auto-release: Comp-dependent Max. Compression: 18dB
Peak Limiting: This setup trims only the very loudest peaks, leaving most of the signal intact. Threshold: -37dB (adjust according to the sound) Ratio: 2:1 or 3:1 Attack: Instantaneous Release: 30 msec Gain: 0dB Soft Knee: Off Gate: Off Comp. Lookahead: -5 msec Max. Compression: Unlimited
Vocal Compression/Spoken Word: This setup compresses the entire dynamic range of the vocal. Whenever there is a signal present, there is some compression taking place. Threshold: Adjust so that the rst bar of the meter comes on even on soft passages. Ratio: 2:1 Attack: 0.1 msec Release: 100 msec Gain: Set to compensate for lost gain. Soft Knee: Off Gate: Off Comp. Lookahead: 0 msec Auto-release: Off Max. Compression: 12dB Backwards Drums & Cymbals: This is a special effect which reverses the volume envelope of cymbals and drums. Threshold: -37dB (adjust according to the sound) Ratio: -1:1 (Neg. Compression enabled) Attack: Instantaneous Release: 200 msec Gain: +19dB Soft Knee: Off Gate: Off Comp. Lookahead: -24 msec Auto-release: Off Max. Compression: Unlimited

Parameter Preferences

Description The Preferences menu allows you to: Toggle the Tooltips On or Off Extra Buffers - Check this box if excessive stuttering occurs when using E-MU PowerFX in your VST Host application. This box should be checked when using Fruity Loops. Render Mode - Induces realtime rendering in applications which do not support realtime rendering (WaveLab, SoundForge).
To Setup & Use E-MU PowerFX:

Setup Cubase LE

1. Launch Cubase LE. 2. Instantiate E-MU PowerFX in an Insert or Aux Send location within Cubase (go to
the EMU folder in VST plug-ins).
3. Press the Insert Edit button
Using any driver other than E-MU ASIO may produce undesirable results when using E-MU PowerFX.
in Cubase to bring up the E-MU PowerFX plug-in window shown on the previous page.
4. Make sure the Insert Enable button
is illuminated, indicating E-MU Power FX is on. The blue Signal Present indicators will be illuminated if E-MU PowerFX is properly patched into a signal path.
5. Drag the desired effects from the Effects Palette to the center Insert strip. 6. Click on the Effect you wish to edit in the center Insert Strip (it will be highlighted in
yellow), then adjust the effects parameters in the right section of the window.
7. You can also select or edit User Presets from the section below the FX parameters.
See the User Preset Section for more information.

Delay Compensation

If you are using Cubase VST 5.1, or another older sequencer without automatic delay compensataion, you will have to insert an E-Delay Compensator into any other audio tracks to keep them time-aligned.
Simply insert an E-Delay Compensator plug-in into the same insert location you used for E-MU PowerFX on any other audio tracks. Thats it.

Automating E-MU PowerFX

E-MU PowerFX can be automated in Cubase LE (or other recording host) just like any other VST effect. When Write Automation is activated in Cubase LE, control changes made in the PowerFX window during playback will be recorded on a special Audio Mix track located at the bottom of the Arrange Window. When Automation Read is activated, the recorded control changes will be played back. To Record PowerFX parameter changes in Cubase LE
1. Add E-MU PowerFX as a Channel Insert. 2. Rewind the song and enable Automation Write by pressing the WRITE button Steinberg Cubasis does not have the control automation feature.
on, illuminating it. (Refers to Cubase LE. If you are using another application, refer to the documentation.)
3. Bring the E-MU PowerFX window to the front and select the Effect you want to
automate. The effect parameters appear in the TV screen. Make sure the blue On button is lit.

6 - Using High Sample Rates Overview
6 - Using High Sample Rates
When operating at 88.2k, 96k, 176.4k and 192k sample rates, the mixer functionality and number of I/O channels are reduced. These changes are summarized in the following tables. All S/PDIF inputs and outputs are disabled at 176.4kHz and192kHz. The number of ADAT channels also decreases at the 88.2k/96k and 176.4k/192k sample rates (due to the bandwidth limitations of the optical components). When using 88.2kHz, 96kHz, 176.4kHz or 196kHz sample rates: Effect processors are disabled. (Output sends & returns are still available.) ADAT is reduced to 4 chan. at 88k/96k, & 2 chan. at 176k/192k. ASIO channels are reduced to 8 stereo ASIO channels at 88k/96k, and 4 stereo ASIO channels at 176k/192k. At 176.4k/192k, the number of physical inputs/outputs is reduced. At the 176.4k & 192k sample rates, S/PDIF optical is disabled. The ADAT optical interface was originally designed to carry 8 channels at a 48kHz sample rate. We use the Sonorus S/MUX standard to encode audio with higher sample rates onto the ADAT light pipe. In this multiplexing scheme, two ADAT channels are used to carry one 88.2k or 96k stream and four ADAT channels are used to carry one 176k or 192k audio stream. In order to use the ADAT interface at these higher sample rates, you must have other equipment that supports the Sonorus S/MUX standard.
E-MU 1820 System at 88.2k/96k (1010 PCI Card & AudioDock)
All outputs remain active at 88.2k/96k, but the number of ADAT channels is reduced from eight to four (see above). There are two possible input congurations when using the E-MU 1820 system at 88.2k/96k as shown in the chart below. Basically, you have the option of using the (4) ADAT input channels or (4) Line Inputs (Line Inputs 2 and 3). At 88.2kHz/96kHz the headphone output parallels the Monitor output and is no longer independently assignable. E-MU 1820 Inputs/Outputs at 88.2kHz or 96kHz Source ADAT S/PDIF 1 S/PDIF 2 Microphone Line 1 Line 2 Line 3 Line 4 out (monitor) Headphone out Total Inputs (ADAT Option) Inputs (Line In Option) Outputs (monitor) 18

Duplicating SMPTE time code
The Sync Daughter Card always generates clean SMPTE from the SMPTE output when reading SMPTE in. This time code is in sync with the incoming SMPTE and can be used to feed other devices in your studio or to clean up old SMPTE tracks. Copying SMPTE code from track to track produces deterioration of the signal with each generation, although one generation of dubbing will probably be OK.
Other Tips for using SMPTE
1. Use ascending time code. Jumps in the code are OK as long as the SMPTE code
jumps forward in time as the tape moves forward in time. A good way to avoid any problems with this is to simply stripe the entire project with SMPTE before you record any other tracks.
2. Allow enough leader. Leave a few seconds between each song to allow SMPTE to
sync up before the song starts. Keep written logs. Keeping written records of song start points and edit cues can save time and avoid wasteful searching through a project that was recorded earlier.

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

SYNC CARD

M A S T E R

RECORD INPUT

R E M O T E

C O N T R O L

DISPLAY TYPE

NORMAL SMPTE BARS

PITCH CONTROL

TAPE LOCATION

HOURS MINUTES SECONDS FRAMES

ADAT In

optional if ADAT sync isn't used
PITCH UP RECORD INPUT 72 104

BARS BEATS SUB BEATS

PITCH DOWN

PITCH MODE

FIXED VARIABLE

ABSOLUTE RELATIVEE

DISPLAY MODE RESET 0

FORMAT TAPE

DROP FRAME 30 FPS

SMPTE IN MIDI IN

29.97 FPS 25 FPS 24 FPS
SMPTE START EXT SYNC OFFSET

73 105

74 106

75 107

77 109

78 110

79 111

80 112

NAME CURSOR

COPY TAPE LOCATION

R MIDI UTIL TEMPO MAPECORD XFADE 7 STUV SAVE SETUP COPY SONG DELETE SONG TO TAPE 8 WXYZ 0 (CHARS)

AUTO-PUNCH PRE-ROLL

POST-ROLL

81 113

82 114

83 115

TAPE OFFSETTRACK DELAY
LOAD SETUP SET LOCATE LOCATE SONGFROM TAPE

DIGITAL I/O GEN SYNC

XLR Connector 22

Zero-Latency Monitoring 41

Inputs & Outputs

(8) Ch. ADAT Optical In (8) Ch. ADAT Optical Out (2) Ch. S/PDIF Digital In (2) Ch. S/PDIF Digital Out (1) MIDI Input & Output (2) 24-bit Bal. Line Inputs (2) 24-bit Bal. Line Outputs
(8) Ch. ADAT Optical In (8) Ch. ADAT Optical Out (2) Ch. S/PDIF Digital Ins (4) Ch. S/PDIF Digital Out (2) MIDI Inputs & Outputs (6) 24-bit Bal. Line Inputs (8) 24-bit Bal. Line Outputs (2) Mic./Line Preamp Inputs (2) Turntable Preamp Inputs (2) Ch, Headphone Outs (4) Computer Speaker Outs
(8) Ch. ADAT Optical In (8) Ch. ADAT Optical Out (2) Ch. S/PDIF Digital In (4) Ch. S/PDIF Digital Out (2) MIDI Ins & 3 MIDI Outs (6) 24-bit Bal. Line Inputs (8) 24-bit Bal. Line Outputs (2) Mic./Line Preamp Inputs (2) Turntable Preamp Inputs (2) Ch, Headphone Outs (4) Computer Speaker Outs (1) Word Clock In & Out (1) SMPTE (LTC) In & Out

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 no load on your computers CPU. The Firewire port provides high-speed connectivity to the Creative NOMAD portable digital audio player, external CD-RW drives and other Firewire compatible devices such as DV camcorders, printers, scanners and digital still cameras. The E-MU 1010 PCI Card also provides eight-channels of ADAT 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 exibility in routing your audio between physical inputs and
E-MU Digital Audio System 1
outputs, virtual (ASIO) inputs and outputs and internal hardware effects and busesno external mixer needed. You can add digital effects, EQs, meters, level controls and ASIO 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 specic purposes such as recording, mixdown, special effect setups or general computer use.

your other gear. 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 enter the serial numbers of the E-MU 1010 PCI Card and AudioDock. These numbers can help EMU Customer Service troubleshoot any problems you may encounterby writing the numbers down now, youll avoid having to open your computer to nd them later on.

Safety First!

To avoid possible permanent damage to your hardware, make sure that all connections are made to the E-MU 1010 card and the AudioDock with the host computers power off. Unplug the computers 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 static. 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: Avoid any unnecessary movement, such as scufng 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 dont 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 youll be connecting the cable in order to discharge any static build-up.

Connect the supplied network-type cable from the 10 BaseT jack on the E-MU 1010 PCI card labeled EXTERNAL to the matching connector labeled Card on the AudioDock. The cable supplied with the AudioDock is specially shielded to prevent unwanted RF emissions.
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.

Software Installation

Installing the E-MU 1010 Drivers
The 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 2000 or Windows XP
The software is not compatible with other versions of Windows.
1. After you have installed your audio card, turn on your computer. Windows
automatically detects your audio card and searches for device drivers.
2. When prompted for the audio drivers, click the Cancel button. 3. Insert the E-MU software Installation CD into your CD-ROM drive. If Windows
AutoPlay mode is enabled for your CD-ROM drive, the CD starts running automatically. If not, from your Windows desktop, click Start->Run and type d:\ctrun\ctrun.exe (replace d:\ with the drive letter of your CD-ROM drive). You can also open the CD and double-click Ctrun.exe located in the CTRun folder.
4. The installation splash screen appears. Follow the instructions on the screen to
complete the installation.
5. Choose Continue Anyway when you encounter the Windows Logo Testing

warning screen.

6. When prompted, restart your computer.
Uninstalling all Audio Drivers and Applications
At times you may need to uninstall or reinstall some or all of the audio card's applications and device drivers to correct problems, change congurations, 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 1010 PCI card entry, or the application entry and then click the

16 Creative Professional

4 - The PatchMix DSP Mixer The Toolbar

The Toolbar

New Session Save Session About PatchMix DSP Session Settings
f Click the buttons in the toolbar to learn about their function.

Open Session

Sync Settings

Show/Hide Effects

Global Prefs

New Session Open Session

Calls up the New Session dialog box. New Session. Calls up the standard Open dialog box, allowing you to open a saved Session. Calls up the standard Save or Save As dialog boxes, allowing you to save the current Session. Toggle button that shows or hides the FX palette. Calls up the Sessions Settings window. Session Settings. Calls up the Global Preferences window. Calls up the SMPTE window. (if Sync Card is installed)

Save Session

Show/Hide Effects Session Settings Global Preferences Sync Settings
About PatchMix DSP Right-Click on the E-MU logo to view the About
PatchMix DSP screen, which provides the software and rmware version numbers and other information.
E-MU Icon in the Windows Taskbar
Right-clicking on the E-MU icon in the Windows taskbar brings up the following window.
Opens the PatchMix DSP Mixer.
Calls the Patchmix DSP help system. Disables the spash screen that appears at boot-up. Restores the default PatchMix DSP and driver settings. Closes the PatchMix DSP background program, disabling use of all audio I/O from the E-MU hardware. Open the PatchMix DSP application to start audio again. E-MU Digital Audio System 17
4 - The PatchMix DSP Mixer The Session

The Session

The current state of the PatchMix DSP mixer (fader settings, effects routingseverything!) 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, 96kHz, or 192kHz, but its complete set of features are only available at 44.1kHz or 48kHz. See Chapter 6 - 96kHz & 192kHz Operation for complete details. When you start a new PatchMix DSP Session, the rst choice you make is to select the sample rate. Once set, you can only easily switch between 44.1k and 48k. You cannot switch between 44/48k and 96k and 192k. This is because the number of mixer inputs and outs change signicantly at these high sample rates. In the case of such drastic sample rate changes, 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, S/PDIF input or the Sync Daughter Card (word clock or SMPTE). If the session is set at 44.1kHz or 48kHz and the external source is coming in at 96kHz, you will be given a warning that the external source requires a new session. PatchMix DSP comes with several session templates to choose from so when you create a new session you can either create a blank session based around a designated sample rate, or select from a list of template starting points. In a PatchMix DSP session the number of strips in the mixer is dynamically congurable. See Pre Fader or Post Fader. This allows you to create only those strips you need up to a maximum number determined by available DSP resources and available inputs.

Volume Control
Controls the output level of the strip into the main/monitor mix bus.

Scribble Strip

These convenient buttons allow you to solo or mute selected channels.
This screen shows a mono strip on the left and a stereo strip on the right.

Scribble Strips

Click inside the scribble strip and type a name of up to eight characters.
4 - The PatchMix DSP Mixer Mixer Strip Creation

Mixer Strip Creation

PatchMix DSP is a dynamically congurable mixer. Each mixer session can contain an arbitrary number of strips up to a limit set by the number of available input sources and available DSP resources. To Add a New Strip:
1. Click on the New Mixer Strip button. See Overview of the Mixer 2. The Assign Mixer Strip Input Dialog appears:
3. Select the desired input to the mixer strip from the following choices: Physical Source: Host - ASIO Source input Host - Direct Sound input Stereo analog or digital card input (Analog, ADAT, S/PDIF) Streaming audio from a software application. Window sound sourcesWAVE, Direct Sound, WDM, CD
Mixer Strip Type Physical: I/O Card In Physical: Dock Mic/Line Physical: Dock In
Function 24-bit monophonic analog input from the 0202 Daughter Card. 24-bit monophonic analog input from the AudioDock. 24-bit stereo analog input from the AudioDock.
Physical: PCI Card S/PDIF 2 channel digital audio from the S/PDIF input on the E-MU 1010 card. Physical: PCI Card ADAT Host: WAVE L & R 2 channel digital audio from the ADAT input on the E-MU 1010 card Direct Sound, WDM, Windows Media (Sound generated or handled by Windows, such as game sound, CD player, beep sounds, etc.). 2 channel digital audio from an ASIO source (software application).
f See Pre or Post Fader Aux Sends on page 31.

Host: From ASIO Out

4. Select Pre-Fader Aux Sends or leave the box unchecked for Post-Fader Aux Sends. 5. Click OK to create a new strip or Cancel to cancel the operation.

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 up to +12dB of gain. High output levels may cause clipping on outboard ampliers or other equipment.
This stereo bar-graph meter reects the digital level at the output of the mixer. The topmost red bar represents 0 dB or a full-scale digital signal. The peaks hold for a moment so that short transients can be monitored. Each bar = 1dB.

MAIN MIX

Monitor Output Level
This control adjusts the monitor output level. Keep in mind that since the monitor level control comes after the Main Output Fader, nothing will be heard from your monitors if the main level is turned down.

Monitor Balance Control

This control sets the relative volume of the stereo monitor outputs and works just like the balance control on your home music system. This control is primarily used to make the volume from each speaker sound equal if you are not sitting exactly in the center of the two speakers.

Monitor Output Mute

This button completely cuts off the monitor output and provides a convenient way to instantly kill all sound without having to re-adjust the monitor level later. When the telephone rings, just hit the monitor mute to cut the noise.
E-MU Digital Audio System 37

5 - Effects Overview

5 - Effects

Overview

PatchMix DSP comes complete with a host of great core DSP effects including Compressors, Delays, Choruses, Flangers and Reverb. Each 32-bit effect has various parameters for editing, as well as factory presets. You can also create and save as many of your own effect presets as you wish. Since the effects are implemented in hardware, they dont place any load on your host computer. This allows your valuable CPU cycles to be used for other applications or software plug-ins. This is a 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 greyed out and cannot be added to the mixer. Complex effects such as reverb, use more DSP resources than say a 1-Band EQ. If you continue to add effects, all of the DSP resources will eventually be used up.

Click here for Edit Menu Click here to Select Presets
E To copy or share User Presets, you must save them as FX Palette effects.

To Select a User Preset

1. Select the FX display in the TV screen. 2. Select the desired insert effect, highlighting it. The effect parameters appear in the TV

screen.

3. Click on the
icon on the preset menu. A drop-down preset list appears.
4. Select a preset from the list.
To Create a New User Preset
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.

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 conrm your action. 4. Click OK to delete the preset or No or Cancel to cancel the operation.

To Rename a User Preset

1. Select the user preset you wish to rename from the user preset menu. 2. Click on the Edit button. A pop-up menu appears. 3. Select Rename. A pop-up dialog box appears asking you to rename the preset. 4. Type in the new preset name, then click OK to rename the preset or Cancel to cancel
the operation. To Overwrite or Save a User Preset This operation allows you to overwrite an existing preset with a newer version.
1. Select the user preset you wish to modify from the user preset menu and make any

changes you wish.

2. Click on the Edit button. A pop-up menu appears. 3. Select Overwrite/Save. The current preset will be overwritten with the new settings.
5 - Effects Core Effects Descriptions
Core Effects Descriptions

1-Band Para EQ

+15dB Boost Width
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.

Center Frequency

Parameter Gain Center Frequency Bandwidth

f You can also type in exact frequencies to a resolution of 1/10 Hz.
Parameter Frequency Left Direction Right Direction
Description Sets the number of Hz that will be added or subtracted with every harmonic in the signal. Sets pitch shift up or down for the left channel. Sets pitch shift up or down for the right channel.

Leveling Amp

The rst compressors developed in the 1950s were based on a slow-acting optical gain cells which were able to control the signal level in a very subtle and musical way. This effect is a digital recreation of the leveling amps of yesteryear. The leveling amp uses a large amount of lookahead delay to apply gentle gain reduction. Because of this delay, the leveling amp is not suitable for applications which require realtime monitoring of the signal. This smooth and gentle compressor is designed to be used in situations where delay does not pose a problem, such as mastering a mix or compressing prerecorded stereo material. Post Gain is the only control on the leveling amp. This control is used to make up the volume lost by the compression. The Compression Ratio is 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 Gain Amplies the signal after it has been compressed to bring up the volume.

Mono Delay 100

The Mono Delay 100 is a monophonic delay line with a maximum delay of 100 milliseconds. Its useful for slapback, doubling and stutter effects when you dont want or need a stereo delay. Stereo signals are summed together before entering the Mono Delay. Very short delay times combined with a high feedback amount can be also used to create monotone robotic-sounding effects.
Feedback HF Rolloff L In L Out

R In Delay Time R Out

Parameter Delay Time Feedback High Freq. Rolloff
Description Sets the length of the delay from 0-100 milliseconds. Sets the amount of delayed signal that will be recirculated through the delay line. Damps high frequencies in the feedback path.

Mono Delay 3000

A delay line makes a copy of the incoming audio, holds it in memory, then plays it back after a predetermined time. Long delays produce echoes, short delays can be used for doubling or slapback effects. Very short delays can be used to produce resonant anging and comb lter effects (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. A High Frequency Rolloff 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.

Description Modeled from a British 8-speaker high power amplier stack. Modeled from a British 2-speaker combo amplier. Modeled from an American, 1950s era, 2-speaker combo amplier. Modeled from an American, 1960s era, 2-speaker combo amplier. Modeled from an American, 1960s era, 4-speaker amplier set. Modeled from a modern era, power amplier stack.

Stereo Delay 100

The Stereo Delay 100 offers a true stereo delay line in that the left and right channels are kept entirely separate from each other. This effect has a maximum delay of 100 milliseconds,perfect for slapback, doubling and stutter effects. Because the left and right channels can have different delay times, you can create a panning effect by setting one delay long and the other short. Very short delay times combined with a high feedback amount can be used to create monotone roboticsounding effects.

Feedback HF Rolloff In

L Delay R Delay Time Time
Parameter Left Delay Time Right Delay Time Feedback High Freq. Rolloff
Description Sets the length of the delay for the left channel from 0-100 ms. Sets the length of the delay for the right channel from 0-100 ms. Sets the amount of delayed signal that will be recirculated through the delay line. Damps high frequencies in the feedback path.

Stereo Delay 1500

The Stereo Delay 1500 offers a true stereo delay line in that the left and right channels are kept entirely separate from each other. This effect has a maximum delay of 1500 milliseconds (1.5 seconds), making it suitable for creating long echoes Because the left and right channels can have different delay times, you can create a panning effects by setting one delay longer than the other. With the feedback turned up you can overdub musical lines one on top of the other.
Description Sets the length of the delay for the left channel from 0-1500 ms. Sets the length of the delay for the right channel from 0-1500 ms. Sets the amount of delayed signal that will be recirculated through the delay line. Damps high frequencies in the feedback path.

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 denes the time it takes for the reected sound from the room to decay or die away. The diagram below shows a generalized reverberation envelope.

Early Reflections

Open the PatchMix DSP Mixer
1. Open the PatchMix DSP mixer by clicking on the
icon on the Windows Taskbar.
f Because unused strips waste DSP resources as well as needlessly complicating the mixer display, you may want to delete the strips you arent using.
To Load the ASIO Direct Monitor Session
2. Click the New Session button,
which is the upper left button above the TV
screen. The following screen appears:
3. From the44k/48k tab, select 44.1 ASIO Direct Monitoring. This Session has all
analog inputs activated with ASIO Direct Monitor Send/Returns as well as a WAVE strip and one pair of ASIO outputs (31& 32) used for monitoring the main output of Cubase.
4. The Session Settings screen appears showing 44.1kHz. Click OK. 5. If you wish to record in stereo, connect an instrument or microphones to inputs
1L/1R. To record in mono, connect to Mic/Line A.

Open Cubase

6. Open Cubase by clicking on the desktop shortcut. 66 Creative Professional
Cubase Audio System Setup Settings
(Options, Audio Setup, System)
The dialog box above shows the correct settings. Make sure your settings match this screen.
E-MU ASIO Control Panel Settings
Click the ASIO Control Panel button to set the ASIO Buffer Latency.
ASIO Buffer Latency should be set to 5 milliseconds or less, depending on the speed of your computers CPU. This control adjusts the amount of delay incurred transferring audio between the Digital Audio System hardware and Cubase.
E-MU Digital Audio System 67
7 - Working with ASIO Basic Recording 7. In Cubase, select File, Open. 8. Open the Cubase song (1820) Cubase Recording Template, which is located
here: (My Computer\Local Disk-C\Program Files\Creative Professional\PatchMix DSP\Recording Templates) This demo contain an empty song, ready for recording.
The Cubase Demo Song Screen
Step 12 - Select Track Arrange Window Return-to-Zero

Transport Bar

Inspector

VST Inputs Panel

Input Selector Switch

Input Meter Switch

VST Channel Mixer 1

VST Master Mixer

Select the Audio Files Folder
Choose the location where your audio les will be stored. Its ususally a good idea to do this step rst as if you were, Putting the tape reels on your recorder.
Go to Options, Audio Setup, Audio Files Folder. ususally preferable for recording audio les.

Digital Mixer

Word Clock
ADAT Optical or AES Digital

ADAT Optical

T - connector
Word Clock Termination OFF

E-MU 1010 CARD SYNC CARD

Word Clock Termination ON
This diagram shows the proper way to connect word clock if you dont have a multi-output word clock generator. The last device in a Word Clock chain should have Termination ON. E-MU Digital Audio System 91
8 - Appendix Useful Information

Useful Information

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
Cables - balanced or unbalanced?
All inputs 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 ne for most applications. If youre having problems with hum and noise or just want the best possible performance, use balanced cables. 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 cables ONLY if you are connecting balanced inputs to balanced outputs.

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 TRS Plug

Balanced XLR Plug

(male)

Balanced cables have one ground (shield) connection and two signal-carrying conductors of equal potential but opposite polarity. There is one hot or positive lead, and a cold or negative lead. At any point in time, both conductors are equal in voltage but opposite in polarity. Both leads may pick up interference, but because it is present both in and out of phase, this interference cancels out at the balanced input connection.

Unbalanced Cables

Unbalanced cables have one conductor and one ground (shield) and usually connect via unbalanced 1/4" phone plugs or RCA phono plugs. The shield stays at a constant ground potential while the signal in the center conductor varies in positive and negative voltage. The shield completely surrounds the center hot conductor and is connected to ground in order to intercept most of the electrical interference encountered by the

cable. Unbalanced cables are more prone to hum and interference than balanced cables, but the shorter the cable, the less hum introduced into the system.

Digital Cables

Dont cheap out! Use high quality optical (ADAT) ber and low-capacitance electrical (S/PDIF) cables when transferring digital I/O to avoid data corruption. Its 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 ber light pipes).

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 preamplier of a condenser microphone. Some direct boxes also use phantom power. Pins 2 and 3 of the AudioDock microphone inputs each carry +48 volts dc referenced to pin 1. Pins 2 and 3 also carry the audio signal which rides on top of the constant 48 volts DC. Coupling capacitors at the input of the AudioDock 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 or this bias could affect the audio headroom.

+48V (grd) 2

Balanced dynamic microphones are not affected by phantom power. An unbalanced dynamic microphone may not work properly, but will probably not be damaged if phantom power is left on. Ribbon microphones should NOT be used with phantom power on. Doing so can seriously damage the ribbon element. Since ribbon microphones are fairly specialized and generally expensive, youll know if you own one. Most microphones are either of dynamic or condenser type and these are not harmed by phantom power.
Appearance Settings in Windows
Adjusting the Performance Options in Windows will improve the screen appearance when moving the mixer around on the screen. To Improve the Appearance Settings:
1. Open the Windows Control Panel. (Start, Settings, Control Panel) 2. Select System. Select the Advanced Settings tab. 3. Under Visual Effects, select Adjust for Best Performance. Click OK.
8 - Appendix Technical Specications

Technical Specications

Background program, disabling 17 Balance Control monitor 37 Balanced Cables 11, 92 Block Diagram, mixer 16 Bypass effect insert 42 send/return insert 34

Index F

flanger 52 frequency shifter 53 leveling amp 54 mono delay mono delay overview 39 palette 39 phase shifter 56 placing into an insert location 24 preset create new 43 delete 44 overwrite 44 rename 44 rotary 56 save in effects palette 40 selecting 40 stereo delay stereo delay stereo reverb 59 using in VST host application 80 vocal morpher 60 E-MU 0202 Daughter Card description 13 installing 4 E-MU 1010 PCI Card description 7 installing 3 E-MU Icon 17 Envelope, reverberation 59 E-Wire 80 Exit PatchMix DSP Services 17 External Clock 14, 20, 90 External Mode, SMPTE 86 External Sync Source 20
Headphone level & specifications 98 output 9 submix, creating a 78 Help System 17 High Frequency Damping, stereo reverb 59 High Frequency Rolloff, mono delay 55 Host Input Display 35 Host Mode, SMPTE 86 Host Output Display 35 Hum, in the audio 93
IEEE Input 86 display 35 level line 11 setting 28 SMPTE 86 specs 94, 97 reduction at high sample rates 61, 62, 63 type mixer strip 22 red color 22 Insert add effect 24 add send 25 add send/return 25, 26, 79 bypass 27, 42 delete 27 menu 25 meter 28 mixer strip 24 solo 27, 43 types 24 Installing disk drive power cable 5 E-MU 0202 daughter card 4 E-MU 1010 PCI card 3 rubber feet 5 sync cables 14 sync daughter card 4 Interface ADAT 7 EDI 12 MIDI 9, 12, 13, 14 required cable 2 S/PDIF 7, 9 SMPTE 14, 85 word clock 90 Invert, polarity 29
Factory Templates 19 Firewire Connector 8 Flanger 52 Flywheel Mode, SMPTE 86 Folder, effects 41 Frame Rates, SMPTE 87 Frequency Shifter 53 Front Panel Connections, Audio Dock 9 Full-Frame Messages 90 FX Display 34 FX Edit Screen 42
Gain, compressor 50 Ground Loop, preventing 93 Ground Lug, turntable 11 Grounding 93

Index L

Label, scribble strip 32 Latency setting E-MU ASIO 67, 71 Latency, monitoring without 26 LED clock source 10 green 9 MIDI activity 10 red 9 sample rate indicator 10 Level Fader 32 Leveling Amp 54 LFO flanger 52 phase shifter 56 vocal morpher 60 Limiter 50 Line Level input/output audiodock 11 I/O daughter card 13 Loss of sync 10 Low Frequency Damping 59
Main bus 33 inserts 37 output fader 37 section 33 Master return level 33 send level 33 Meter insert 27 main output 37 setting input levels using 28 Microphone Preamps 9 MIDI I/O jacks 0202 Daughter Card 13 AudioDock 9, 12 input indicator 10 jacks 9, 12 settings 20 time code 90 Mini-Phone Outputs 12 Mixer block diagram 16 overview 15 strip 22 aux send 30 fader 32 insert 24