E-MU 1820M, size: 4.1 MB
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Emu 1820M PreAmp Noise/Hum??60Hz
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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 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 1820M/1820/1212M PCI Digital Audio System 9
outputs, virtual (ASIO/WAVE) inputs and outputs and internal hardware effects and busesno 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 specic purposes such as recording, mixdown, jamming, special effect setups, playing games, watching DVDs, or general computer use.
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 1820 System
The E-MU 1820 includes the AudioDock, which is a half rack-space, audio interface. The AudioDock adds the following input and output capabilities to the system: two mic/line inputs with TFPro preamps, 6 balanced line level analog inputs, an RIAA stereo turntable preamp, 8 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. Combined with the digital I/O on the 1010 PCI card, you have a total of 18 inputs and 20 outputs! Of course, professionalquality, 24-bit A/D and D/A converters with automatic DC blocking are used throughout.
E-MU 1820M System
The E-MU 1820M system includes the AudioDockM, and is a no compromise, mastering-grade system, which includes all the features of the 1820 system. The 1820M system is distinguished by the addition of ultra-high performance 24-bit/192kHz A/D - D/A converters which deliver an unbelievable 120dB dynamic range. The Sync Daughter Card comes standard with the 1820M system and can be purchased as an optional upgrade to the 1820 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. Youll want to keep up with the latest software and options for your E-MU digital audio system. You can nd all of this, plus other helpful information, at the E-MU Website: http://www.emu.com.
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)
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 Digital Audio System
4 - The PatchMix DSP Mixer 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 44.1k, 48k, 88.2k, 96k, 176.4k or 192k, but its complete set of features are only available at 44.1kHz or 48kHz. See Chapter 6 - Using High Sample Rates for complete details. Once the sample rate is set, you can only easily switch between 44.1k and 48k. You cannot switch between 44/48k and the higher rates of 88k/96k/176k/192k. This is because the number of mixer inputs and outputs changes 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. If the session is set at 44.1kHz or 48kHz and the external source is coming in at 96kHz (for example), 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 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. This allows you to create only those strips you need up to a maximum number determined by available DSP resources and available inputs.
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.
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:
Physical input (Analog/SPDIF/ADAT). Host Input (Direct Sound, WAV, ASIO source)
f The Input Type will turn RED if the input is not available. (The AudioDock may be disconnected.)
f Physical input strips are shown with BLUE text. f Host input strips are shown with WHITE text.
You can drag and drop effects from the Effects Palette or Right-click to insert a Physical or ASIO Send or Send/Return A Peak Meter, Trim Control or Test Signal can also be inserted by Rightclicking.
These controls allow to you position the channel in the stereo sound eld. Dual controls on stereo strips allow you to position each side independently.
Channel Volume Control
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.
Controls the output level of the strip into the main/monitor mix bus.
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.
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. You must create a strip for each mono or stereo audio input, and for each ASIO stream you wish to use in your software application. This is important because outputs will not appear in your software application until you have created ASIO strips in PatchMix. Host refers to a computer application such as Cubase. Physical refers to hardware input or output such as a output jack. To Add a New Strip:
There are several utilities to help you organize your effects presets. To Create a New Preset Category You can create your own category folders to help organize your effects presets.
1. Left-click on the New Folder icon at the top of the Effects Palette. A pop-up dialog
box appears asking you to Enter the Name of the New Category. Alternatively, you can Right-click over an Effects Folder, which calls a pop-up dialog box with the option to Create New Category.
2. Type in a name for your new folder. 3. Click OK to create a new folder or Cancel to cancel the operation.
To Delete an Effect Category or Preset
1. Right-click on the category folder you wish to delete. A pop-up selection box
2. Select Delete Category. A popup dialog box appears warning you that this action
will delete all presets in the folder.
3. Click OK to delete the folder or Cancel to cancel the operation.
To Rename an Effects Category
1. Right-click on the category folder you wish to rename. A pop-up selection box
2. Select Rename Category. A pop-up dialog box appears, asking you to Enter New
3. Click OK to rename the folder or Cancel to cancel the operation.
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. To Import Core FX Presets This option imports complete folders of Core FX presets into the E-MU PatchMix DSP folder (normally located here: C:\Program Files\Creative Professional\E-MU PatchMix DSP\Core Effects). 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 Browse for Folder 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. To Import FX Category Folders This option imports complete category folders of FX Chains into the E-MU PatchMix DSP folder (normally located here: C:\Program Files\Creative Professional\E-MU PatchMix DSP\Effect Presets). 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.
2. Select Import FX Category. The Browse for Folder 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. To Export your Core FX Presets This option exports your Core FX presets to a folder of your choice.
2. Select Export FX Library. The Browse for Folder 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.
To Export your FX Category Folders This option exports a single category of FX chains to a folder of your choice.
2. Select Export FX Category. A pop-up dialog box appears asking you to Choose the
FX Category to be exported.
3. Choose the desired FX Category to export. Press OK to continue or Cancel to
cancel the operation.
4. The Browse for Folder 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.
5 - Effects FX Edit Screen
FX Edit Screen
Click on an FX Insert to display the parameters for that effect. If an insert effect is not selected, the FX display will read No Insert. Most effects have a wet/dry mix parameter to control the ratio of effect-to-plain signal. The wet/dry setting is stored with the FX preset. The effect parameters vary with the type of effect. Generally if an effect is placed in an Aux Send, the wet/dry mix in the effect should be set to 100% wet since the Aux Return amount controls how much effect is applied. The User Preset section is located at the bottom of the FX Edit screen. User presets are variations of the main effect and can be edited, deleted, renamed or overwritten as you wish.
Effects Display View Button Effect Location Effect Bypass & Solo Buttons Wet/Dry Mix Control
1. Click on the Effect (in the Insert section) 2. Click the Bypass button in the TV display.
1. Right-click over the Insert you want to bypass (in the Insert section). A pop-up
2. Select Bypass Insert from the list of options. The insert effect name will gray-out
to indicate that the insert effect is bypassed. To Solo an Insert: Inserts can also be soloed. Solo bypasses all the other inserts in the strip and allows you to hear only the soloed effect. This feature is very useful when adjusting the effect parameters. Method #1
1. Click on the Insert Effect (in the Insert section). 2. Click the Solo button in the TV display. E-MU Digital Audio System 59
1. Right-click over the Insert Effect you want to Solo (in the Insert section). A pop-up
2. Select Solo Insert from the list of options. The other Insert Effect names in the
strip will gray-out to indicate that they are bypassed. To Bypass ALL All the inserts in a strip can be bypassed with a single command.
1. Right-click over any Effect in the Insert section. A pop-up menu appears. 2. Select Bypass All Inserts from the list of options. All the insert names will be
grayed-out to indicate that they are bypassed. To Un-Bypass ALL All the inserts in a strip can also be un-bypassed with a single command. This command works even if only some of the effects are bypassed.
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. Range:.01Hz to 24kHz Sets pitch shift up or down for the left channel. Sets pitch shift up or down for the right channel.
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. Range 0dB to 36dB
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 ve Lite Reverbs can be used at once. 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.
After a short pre-delay period, the echoes from the closest walls or ceiling are heard. These rst echoes, or Early Reections, vary greatly depending on the type of room. Some time after the early reection cluster ends, the actual Reverberation (a dense cloud of complex wall reections) 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 reections and the reverberation cloud. High frequency energy tends to fade away 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 reective and have less high frequency damping. Rooms 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 boominess of the room. Parameter Decay Time HF Decay Factor Description Sets the reverb decay time. Range: 0% to 100% Sets the rate at which high frequencies die away. The high frequencies last longer as the percentage is increased. Range: 0% to 100% Sets the rate at which low frequencies die away. The low frequencies last longer as the percentage is increased. Range: 0% to 100% Sets the volume of the initial wall reections. Range: 0% to 100% Sets the amount of scattering of the early reections and the reverberation cloud. Range: 0% to 100%
LF Decay Factor
Early Reections Reverberance
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 efcient 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 anging and comb 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 ltering produced by the delay line. Comb ltering: See page 70. 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.
Feedback HF Rolloff L In L Out
R In Delay Time R Out
Parameter Delay Time
Sets the length of the delay in milliseconds. (.01ms. minimum increment between settings) Mono Delay 100 Range: 1 millisecond to 100 milliseconds Mono Delay 250 Range: 1 millisecond to 250 milliseconds Mono Delay 500 Range: 1 millisecond to 500 milliseconds Mono Delay 750 Range: 1 millisecond to 750 milliseconds Mono Delay 1500 Range: 1 millisecond to 1.5 seconds Mono Delay 3000 Range: 1 millisecond to 3 seconds Feedback High Freq. Rolloff Sets the amount of delayed signal that will be recirculated through the delay line. Range: 0% to 100% Damps high frequencies in the feedback path. Range: 0% to 100%
A phase shifter produces a 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 1970s 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 xed multi-notch lter is created.
Parameter LFO Center Feedback LFO Rate LFO Depth Waveform LFO L/R Phase
As input signals exceed the Threshold, the rightness character in the bargraph is lit, and successive characters are lit for each approximately 3dB in gain reduction imposed by the compressor on the input signal. Because this is a compression meter and not a level meter, the same input signal level will show widely varying meter readings depending on the setting of the Ratio parameter.
Sets the ratio of output signal to input signal levels, selectable in 16 steps from 1:1.1 to 1:INFINITY. When Neg Compression is set to Enabled, the range of compression ratios extends beyond INFINITY to encompass negative compression ratios from 1:-100 down to 1:-1, which can be useful for applications like ducking and other special effects. See the discussion of the Neg Compression parameter on page 96.
0dB 1.1:1 1.5:1 2:1 3:1 10:1 :1 8
f Tip: A ratio of Innity:1 combined with high threshold and fast attack/release results in an effective peak limiter.
Sets the amount of time that the compressor's level detector will take to respond to an increase in signal level. The Attack range is adjustable from Instantaneous (essentially a peak detector that follows individual samples) to 10 seconds (useful for long-term leveling or automatic mixing applications.)
Sets the amount of time that the compressor's level detector will take to respond to a decrease in signal level. The fastest Release time is 100 microseconds, useful for some special effects but highly prone to distortion; more typical release times are in the range of 70 milliseconds to 1 second. Release times up to 10 seconds are available for longterm leveling or automatic mixing applications. When the Auto-release parameter is in its signal-dependent settings, the Release time shown represents the shortest possible release time. In Auto-release modes the displayed Release time will be automatically extended depending on the dynamics of the input signal.
E-MU Digital Audio System 91
Sets the compressor's output gain in dB, from +60dB boost to-60dB cut. This control follows all of the other elements in the compressor's signal path, so positive gain boost can be used to make up for the gain reduction normally applied to signals above the compression threshold. Alternatively, negative gain cut can be used to make up for the gain increase that is applied to signals below the threshold in Soft Knee mode. Auto Makeup Mode: When adjusted downward past the -60dB cut, the Gain parameter begins operating in Auto Makeup mode. Auto Makeup mode is used to compensate for the drop in output level normally resulting from the gain reduction actions of the Threshold and Ratio parameters. Auto Makeup makes it much easier to adjust these parameters since there is no need to switch back and forth to the Gain parameter in order to perform the gain compensation manually. Auto Makeup looks at the gain reduction implied by the setting of the Threshold and Ratio parameters and automatically applies a complementary gain increase so that an ideal 0dB input signal results in a 0dB - or lower - output signal. In this mode, indicated by the Threshold legend, the Gain parameter adjusts the output level from that 0dB input signal to fall anywhere in the range of 0dB down to -60dB.
E Note: Its easier to use E-MU PowerFX instead of E-Wire if you just want to use the hardware effects. (E-Wire was the precursor to E-MU PowerFX.) However, E-Wire can be very useful because it allows you to route VST inserts or Sends to Physical Inputs and Outputs via PatchMix DSP.
E-Wire VST plug-in
Send to Strip
Return to VST
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. 106 Creative Professional
To Setup and use E-Wire:
Setup PatchMix DSP
1. Open PatchMix DSP application. 2. Insert an ASIO Input mixer strip into PatchMix DSP. (Alternately, you can select
New Session, select E-Wire Example and skip to step 6.)
3. Mute the strip or turn the Fader all the way down. 4. Insert an ASIO Send plug-in into one of the inserts on your ASIO strip. 5. Name your ASIO strip as an E-Wire strip. 6. Insert the desired PatchMix DSP effects into slots above the ASIO Send. 7. Save the Session.
8. Launch Cubase LE. 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
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. Thats it.
As audio is transferred back and forth between the VST host application and the E-MU sound hardware, a delay in the audio stream is incurred. Normally this delay is compensated for automatically by the host application, but not all VST host applications support this automatic compensation.
A host will support PowerFX and E-Wires plug-in delay compensation if it supports the SetInitialDelay feature of the VST 2.0 specication.
E-MU Digital Audio System 107
Currently automatic delay compensation is supported by the Steinberg 2.0 family (Nuendo 2.x, Cubase SX 2.0, Cubase LE 2.0,), Magix Samplitude 7.x, and Sonar (using the Cakewalk VST adapter 4.4.1), but not by Steinberg Cubase VST 5.1 and Cubasis. The E-Delay Compensator utility plug-in is used to manually compensate for the transfer delay for hosts that DO NOT support plug-in delay compensation. The E-Delay Compensator plug-in is used to delay the dry tracks (tracks without a PowerFX or E-Wire as an insert effect) or auxiliary (send) channels. For each dry track or send, add an E-Delay Compensator plug-in to re-align the track. The E-Delay Compensator is automatic and requires no user interaction to operate. For example, consider a Cubase VST session with two audio tracks. If PowerFX or E-Wire is applied as an insert effect to the rst audio track, but not to the second, the 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.
Why use SMPTE?
SMPTE sync, although well over 30 years old, has the advantage of being able to be recorded as an audio track. This allows it to be used with virtually any kind of recording equipment from tape recorders to computer-based digital audio recorders. You can even buy phonograph records with a SMPTE stripe! SMPTE was designed in the days when tape dropout was a common occurrence and so it was designed to convey absolute location information. Since each frame of SMPTE code provides its own unique identication, it provides the ability for a receiving device to recover from data dropout. In addition, edits can be performed in the middle of a song with just a few seconds of pre-roll before the punch-in point. SMPTE is also standardized, which means that code generated on different makes of equipment will be compatible with each other. SMPTE also has fairly good resolution, especially at the subframe level. Youll be happy to know that the Sync Daughter Card resolves to the subframe level. The chart below shows subframe accuracy at the three frame rates. SMPTE Subframe Resolution
Frames-per-second 24 fps 25 fps 30 fps Resolution.521 mS.500 mS.417 mS
Printing SMPTE to a track is called striping (as in stripe). SMPTE time code is recorded on an unused audio track of another recorder, then played back into the Sync Daughter Card. The Sync Daughter Card passes the location information on to the host computer as MTC quarter-frame data to be used by an application such as an audio recorder or sequencer. SMPTE is usually recorded at about -3 VU on semi-pro gear, -10 VU on professional gear and 0 VU on video gear. Experiment to nd the optimum levels. When printing to a time code track of a video deck, be careful. The time code playback head locations on video decks are not standardized and can cause gross timing errors. Time code which is striped on an audio track will always be in sync with the picture. SMPTE code is traditionally recorded on the right channel of a video recorder.
Avoiding SMPTE problems
Problems in reading SMPTE time code can often be related to poor quality code on the tape. Poor quality code can be caused by a number of problems, the most common being dirty or misaligned heads, amplier clipping, or too many generations of audio dubbing. Other problems can be caused by running the SMPTE signal through signal processing devices such as Limiters, Reverbs, Harmonizers, etc. (Dont laugh, it has been done!) In fact, many video decks have built in AGCs (Automatic Gain Controls) which will ruin the SMPTE signal if the input level is too high. Always check playback to insure that the time code is usable. In general, no signal processing should be used on the SMPTE signal. SMPTE code is delicate and should be treated as such.
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.
United States Compliance Information
FCC Part 15 Subpart B Class B using: CISPR 22 (1997) Class B ANSI C63.4 (1992) method FCC Site No.90479
Canada Compliance Information
ICES-0003 Class B using: CISPR 22 (1997) Class B ANSI C63.4 (1992) method Industry of Canada File No.IC 3171-B
European Union Compliance Information
EN55024 (1998) EN55022 (1998) Class B EN61000-3-2 (2001) EN61000-3-3 (1995 w/A1:98)
Australia/New Zealand Compliance Information
AS/NZS 3548(1995 w/A1 & A2:97) Class B EN55022 (1998) Class B
Japan Compliance Information
VCCI (April 2000) Class B using: CISPR 22(1997) Class B VCCI Acceptance Nos. R-1233 & C-1297
Attention for the Customers in Europe
This product has been tested and found compliant with the limits set out in the EMC Directive for using connection cables shorter than 3 meters (9.8 feet).
If static electricity or electromagnetism causes data transfer to discontinue midway (fail), restart the application or disconnect and connect the Firewire cable again.
Auto-Wah 66 Aux Bus 46 Auxiliary Effects Assignment 52 Auxiliary Returns 52 Auxiliary Sends 46 used as extra mix busses 52
0202 Daughter Card PCI Card 19 1-Band Para EQ 63 1-Band Shelf EQ 63 1-Time Jam Sync, SMPTE 118 3-Band EQ Volt Phantom Power 22, 126 4-Band EQ 65 88kHz/96kHz Sample Rate 111 96kHz Sample Rate 111 Background program, disabling 29 Backward Cymbal Effect 99 Balance Control, monitor 53 Balanced Cables 24, 125 Band Cut Filter 89 Band Pass Filter 89 Block Diagram, mixer 28 Bypass effect insert 59 send/return insert 50
Category create new preset 57 delete effects 57 rename effects 57 CDs, playing 36 Chorus 67 using freq. shifter 71 Clicks & Pops, in the audio 19, 122 Clipping Indicators 22 Clock, external 26, 32 Comb Filter 70 Compressor 67 RFX 90 Connecting, AudioDock 15 Connectors, interface 12 Core Effects descriptions 63 listing 62 Core FX Presets, importing/exporting 58 Cross-over, creating with multimode filter 86
A/D - D/A Converter Type 1212 system system 130 1820M system 127 ADAT Optical at 96kHz & 192kHz 111 connection example 124 input/output connector 19 ADSR, reshaper effect 84 Advanced Parameters, RFX compressor 92 AES/EBU to S/PDIF Adapter 125 Ambience Reduction, using reshaper effect 83 Analog I/O 0202 Daughter Card 20 AudioDock 24 Appearance, improving 126 ASIO direct monitor 41 send 38 Attack curve, reshaper 85 threshold, reshaper 84 Attack, compressor 68 AudioDock front panel indicators 23 inputs/outputs 21 installing power connector 15 rack mounting 16 Auto Makeup, RFX compressor 92 Auto Volume Pedal, using reshaper effect 83 Automating PowerFX 103 Auto-Release, RFX compressor 95
Damping, high frequency 73, 78 Decay Time, lite reverb 73 Decay Time, reverb 78 De-esser, creating 100 Delete folder 57 FX user preset 61 mixer strip 37 Diffusion 78 Digital Cables 126 Digital Interface, S/PDIF 19 Direct Sound Source 36
Distortion 69 Doppler, effect using Rotary 75 Drivers, installing 17 Drop-frame, SMPTE 119 Drum Punch 98 Ducker 97 DVD, playing in 5.1/7.1 surround 37 Dynamic Range 127, 130, 133
Echo, creating 74 E-Delay Compensator 107 Edge, distortion 69 EDI Connector 19, 25 Effects 1-band para EQ 63 1-band shelf EQ 63 3-band EQ 64 4-band EQ 65 auto wah 66 chorus 67 compressor 67 create new folder 57 creating robot voice 74 descriptions 63 display screen 50 distortion 69 edit 56 flanger 70 frequency shifter 71 gate 80 leveling amp 72 lite reverb 73 mono delays 74 multimode eq 86 overview 55 palette 55 phase shifter 75 placing into an insert location 38 preset create new 60 delete 61 overwrite 61 rename 61 recording 57 reshaper 83 RFX compressor 90 rotary 75 selecting 56 stereo delays 77 stereo reverb 78 using in VST host application 101 vocal morpher 79 Effects, order of 57 E-MU 0202 Daughter Card
description 20 installing 14 E-MU 1010 PCI Card description 19 installing 13 E-MU Icon 29 Envelope, reverberation 73, 78 E-Wire 106 Exit PatchMix DSP Services 29 Exporting Core FX & FX Insert Chains 58 External Clock 26, 32, 122, 124 External Mode, SMPTE 118 External Sync Source 32 Extra Buffers 102
Factory Templates 31 Firewire Connector 20 Flanger 70 Flywheel Mode, SMPTE 118 Frame Rates, SMPTE 119 Frequency Shifter 71 Front Panel Connections, Audio Dock 22 Full-Frame Messages 122 FX Edit Screen 59 FX Insert Chains 56 FX Insert Chains, importing/exporting 58
Gain Reduction Meter, gate effect 82 Gain, compressor 68 Gate, RFX compressor 94 Ground Loop, preventing 126 Ground Lug, turntable 24 Grounding 126 GS Wavetable Software Synth 135
Headphone level & specifications 131 output 22 Headphones, using with the Help System 29 High Frequency Damping, stereo reverb 78 High Frequency Decay Factor, lite reverb 73 High Frequency Rolloff mono delays 74 stereo delays 77 Highpass Filter 87 Hold Time, reshaper effect 84 Host Input Display 51 Host Mode, SMPTE 118 Host Output Display 51 Hum, in the audio 126
Neg Compression, RFX compressor 96 New mixer strip 36 session 29, 30 at 176k/192k 115 Noise Gate 80 Notes, Tips & Warnings 10
OpAmp Type 1212 system system 130 1820M system 127 Optical Cables 126 Order of Effects 57 Output fader, main 53 level line 24
Quarter-Frame Messages 122
Rack Mounting, Audio Dock 16 Ratio compressor 68 RFX compressor 91 Recording Effects or Recording Dry 57
Red Strip 35 Reducing Noise 126 Release Curve, reshaper effect 85 Release Time, gate effect 81 Release, compressor 68 Render Mode 102 Reshaper 83 Reverb, envelope 73, 78 Reverberation 78 RFX Compressor 90 RJ45 Connector 15 Robot Voice Effects, creating 77 Rotary, effect 75 Rubber Feet, installing on Audio Dock 16 Rumble Filter, using multimode filter 86
S/MUX 111 S/PDIF cables 126 inputs and outputs 19 optical 22 S/PDIF to AES/EBU Adapter 125 Sample Rate 96kHz & 192kHz 111 indicator LEDs 23 setting 30 Save FX Insert Chains 56 session 31 user effect preset 60 Scribble Strip 48 Send /return insert 39, 40, 41 bypass or solo 50 auxiliary 46 insert 39 Send/Return Levels 49 Session 30 creating 176k/192k 115 creating new 30 path 31 templates 31 at 176k/192k 115 Setting Up the E-MU Digital Audio System 11 Settings I/O 33 input level 22 MIDI 32 system 31 Sibilance, reducing 100 Sidechain creating a de-esser 100 creating a ducker 97 Sidechain Effects 52
routing 46 Signal generator, insert 44 Signal Level Indicators LEDs 22 meters 53 Signal Level, increasing 20, 24 Smooth Bass Guitar Level 98 SMPTE 117 background 119 example connection 121 Soft Knee, RFX compressor 92 Software Installation 17 Solo button 48 insert 59, 60 send/return insert 50 Specifications 1820 System 130, 133 1820M & 1212 Systems 127 Start Time, SMPTE 118 Stereo Delays 77 Stereo Reverb 78 Strip add new 36 input type 35 mixer 35 Striping SMPTE 119, 120 Subwoofer Filter, using multimode filter 86 Surround Sound channel chart 37 playback 37 Sync Daughter Card, description 26 Sync/Sample Rate Indicators 52 Synchronization ADAT example 124 source 32 System Settings 31 System Volume Control 53
T-Connector, word clock 123 Templates, session 31 Termination, word clock 123 Threshold gate effect 81 RFX compressor 91 Threshold, compressor 68 Time Code MIDI 122 SMPTE background 119 SMPTE conversion 117 Toggle Tooltips 102 Toolbar, overview 29 TRS Plugs & Jacks 125 Turntable Inputs 24
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