Creative Emu 1820m Manual

							7 - Appendix
SMPTE Background
E-MU 1820M/1820/1212M PCI Digital Audio System 121
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.
The Sync Card should not be used as both the SMPTE and word clock master. Word 
Clock is generated by the Digital Audio System and NOT by the software application 
(Cubase). SMPTE is not locked to Word Clock inside the Sync Card—they are 
completely independent.
SYNC
CARD 1010
CARD
ADAT Optical 
carries embedded
word clock 
ADOCK
ADAT
In
LT C
BRC
ADAT
REWINDFAST FWDSTOPPLAYRECORDEJECT
MTC
PatchMix DSP set to 
ADAT Sync
1
33
65
972
34
66
983
35
67
994
36
68
1005
37
69
1016
38
70
1027
39
71
1038
40
72
104 RECORD
INPUT
9
41
73
10510
42
74
10611
43
75
10712
44
76
10813
45
77
10914
46
78
11015
47
79
11116
48
80
112 RECORD
INPUT
17
49
81
11318
50
82
11419
51
83
11520
52
84
11621
53
85
11722
54
86
11823
55
87
11924
56
88
120 RECORD
INPUT
25
57
89
12126
58
90
12227
59
91
12328
60
92
12429
61
93
12530
62
94
12631
63
95
12732
64
96
128 RECORDINPUT
TRACK 1-32 TRACK 33-64 TRACK 65-96 TRACK 97-128
AUTO INPUT ALL SAFE ALL INPUT ALL CLEAR
GROUP  1 GROUP  2 GROUP  3 GROUP  4 SET GROUP LOCATE SET LOCATELOCATE 0 SET LOCATELOCATE SONGLOAD SETUP
FROM TAPE COPY SONGDELETE SONGSAVE SETUP
TO TAPE MIDI UTIL PITCH MODE PITCH DOWNPITCH UP
NAME
SMPTE IN MIDI INFIXED VARIABLEHOURS MINUTES SECONDS FRAMES CENTS
PITCH MODE
PITCH CONTROL TAPE LOCATION
CURSOR
AUTO-PUNCHSMPTESMPTE START
OFFSETEXT SYNC DISPLAY MODE RESET 0 FORMAT TAPEDISPLAY TYPE EJECT
PRE-ROLL POST-ROLL
LOOP TAPE OFFSETTRACK DELAY
DIGITAL I/O
AUTO PLAY
REHEARSE GEN SYNC EDIT COPY TAPE
LOCATION
7
STUV8
WXYZ0
(CHARS)
4
JKL5
MNO6
PQR
1
ABC2
DEF3
GHI TEMPO MAPRECORD XFADE
NORMAL
SMPTE
BARS
BARS BEATS SUB BEATSABSOLUTE
RELATIVEE
DROP FRAME
30 FPS
29.97 FPS
25 FPS
24 FPS
MASTER  REMOTE  CONTROL
RECORD PLAY STOP FAST FORWARD REWIND
ADAT 9-pin
optional
if ADAT
sync isnt
used
Cubase 
						

							7 - Appendix
MIDI Time Code (MTC)
122Creative Professional
MIDI Time Code (MTC)

MTC and SMPTE do 
NOT synchronize at the 
sample rate and are not 
locked to word clock in 
any way. 
SMPTE and MTC are used 
to synchronize music but 
do not have the required 
resolution to sample-lock 
digital audio. MIDI time code is basically SMPTE time code adapted to the world of MIDI. MTC 
specifies “absolute” location information in hours:minutes:seconds:frames, just like 
SMPTE. There are two main kinds of messages in MTC: Full-frame messages and 
Quarter-frame messages.
Full-frame messages are ten bytes long and are sent when SMPTE start, stops, or 
relocates. Full-frame messages contain the entire SMPTE number of, hours, minutes, 
seconds, frames, as well as the SMPTE type: 24fps, 25fps, 30fps non-drop, 30fps drop.
Quarter-frame messages are sent at each quarter of a SMPTE frame and only carry 1/8th 
of the SMPTE time message. Quarter-frame messages require two entire SMPTE frames 
to send the complete time stamp (h:m:s:f). Timing accuracy is maintained as long as the 
quarter-frame messages continue to come in at a constant rate.
To Enable MTC:
MIDI Time code disables the use of MIDI port 2 on the back panel of the AudioDock.
1.Open Session Settings from the toolbar.
2.Select the MIDI tab and choose Sync Card/MTC from the MIDI options.
3.Click OK to close the window.
Since it is important to have a stable timing reference for your song or sequence, we 
have given MTC its own MIDI output port on the Sync Daughter Card. This ensures that 
the timing information will not be affected by other MIDI data on the line.
Word Clock In/Out
fWord clock, ADAT and 
S/PDIF synchronize at the 
sample rate and are used 
to transfer digital data 
between machines.Word clock provides a standardized means of synchronizing multiple digital audio 
devices so that data can be transferred digitally. In order to digitally transfer from one 
device to another, the two devices MUST be synchronized. Clicks and pops in the 
audio will result when transferring digital audio which is not synchronized.
The E-MU 1010  PCI card can be externally clocked from the ADAT input, S/PDIF input 
or from the Sync Daughter card (if installed). In a digital studio, all digital devices in 
the system should run off the same master Word Clock. 
To Synchronize PatchMix DSP to an External Clock Source:
1.Make sure an external clock source is connected to the E-MU Digital Audio System 
hardware via the word clock, ADAT or S/PDIF input.
2.Open the Session Settings dialog box.
3.Under the System tab, select External Source, then select either word clock, ADAT 
or S/PDIF.
4.Press OK to close the dialog box.
5.Check the Sync section of PatchMix DSP to verify that the Locked indicator is 
illuminated.
Devices can be connected in daisy chain fashion (word clock out connected to the next 
unit’s word clock in) or in parallel for one or two devices, but professional digital 
studios normally use a master word clock generator or “House Sync” with a distribution 
system so that each device receives a phase-coherent and jitter-free word clock. 
						

							7 - Appendix
Word Clock In/Out
E-MU 1820M/1820/1212M PCI Digital Audio System 123
Word Clock In: Receives word clock (sample clock) from another digital device such as 
a digital video deck, digital recorder or digital mixer.
Word Clock Out: Sends word clock (sample clock) to another digital recorder. Word 
clock is always output, whether it is generated by the internal clock or passed through 
from the word clock input.
75Ω On/Off: Termination for the word clock input can be switched on or off in the 
Sync Card menu of the PatchMix DSP application. Normally word clock termination 
should be left on. If you have problems with a weak word clock signal, try turning termi-
nation off. See W
ord Clock Termination.
The diagram below shows the proper way to connect and terminate a serial word clock 
chain. Using a BNC “T” connector ensures that word clock is precisely in phase for both 
devices. The middle device has termination turned Off and the last device in the word 
clock chain has termination turned On.
House Sync
Generator
Digital
Device 1
Digital
Device 2
Digital
Device 3
Digital
Device 4
A master word clock generator is preferable for larger digital setups.
Word Clock
Word Clock
Word Clock Termination ON Word Clock Termination OFF
SYNC CARD E-MU 1010 CARD
IN
Digital Mixer
ADAT Optical
or AES Digital
ADOCK
IN
T - connector
AES ADAT Optical
This diagram shows the proper way to connect word clock if you don’t have a multi-output 
word clock generator. The last device in a Word Clock chain should have Termination ON.  
						

							7 - Appendix
Getting in Sync
124Creative Professional
Getting in Sync
Whenever you connect external digital audio devices together, you need to be aware of 
how they are synchronized to each other. Simply connecting digital out to digital in 
doesn’t guarantee that two digital devices are synced, even if audio is being passed. 
Unless you have set one to be the Master and the other a Slave, they are probably NOT 
synchronized and the quality of your audio will suffer.
S/PDIF and ADAT are two commonly used digital audio formats. Both these digital 
formats carry an embedded word clock which can be used to synchronize the digital 
equipment. You must enable “External Clock” on the slave device to have clock sync!
The diagrams below show two ways to synchronize an external A/D - D/A converter to 
the E-MU Digital Audio System using the ADAT lightpipe connection. 
In the first example, only the A/D converters on the external device are being used. Only 
one lightpipe is needed as long as PatchMix is set to receive its word clock signal from 
the external device. The external A/D is the Master and the E-MU DAS is the Slave.
In the second example a second lightpipe is used to supply “embedded word clock”, as 
well as eight channels of audio to the external A/D - D/A. The external device MUST be 
set to receive external clock via ADAT or the units will not be synchronized. The E-MU 
Digital Audio System is the Master and the external A/D - D/A is the Slave.
12345678
EXTERNAL
Set External Device to receive:  
External ADAT Sync
ADAT Out
ADAT In
PatchMix DSP supplies Master Clock
(via ADAT)
12345678
EXTERNAL
Set PatchMix DSP to receive:  
External ADAT SyncADAT Out
External Device supplies Master Clock
(via ADAT)
This lightpipe carries an
embedded clock signal
& eight channels of audio. The lightpipe carries eight
channels of audio data and
an embedded clock.
This lightpipe carries eight
channels of audio data.
MasterSlave Master
External A/D - D/A Converter
External A/D - D/A Converter
Slave 
						

							7 - Appendix
Useful Information
E-MU 1820M/1820/1212M PCI Digital Audio System 125
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.
Cables - balanced or unbalanced?
All inputs and outputs on the E-MU Digital Audio System are designed to use either 
balanced or unbalanced cables. Balanced signals provide an additional +6dB of gain 
on the inputs and are recommended for best audio performance, although unbalanced 
cables are fine for most applications. If you’re having problems with hum and noise or 
just want the best possible performance, use balanced cables.

WARNING: Do NOT 
use balanced audio 
cables when connecting 
balanced outputs to 
unbalanced inputs. 
Doing so can increase 
noise level and introduce 
hum. Use balanced 
(3-conductor) cables 
ONLY if you are 
connecting balanced 
inputs to balanced 
outputs.Balanced Cables
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 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.
Output Input
12
321
31 = Ground/shield
2 = 
Hot (+)
3 = 
Cold (-)
Tip = Hot (+) Sleeve = Ground
Ring = Cold (-) 
Tip = Signal Sleeve = Ground
Balanced 1/4” 
TRS Connectors Balanced XLR 
Connectors
Unbalanced 1/4” 
Connectors 
						

							7 - Appendix
Useful Information
126Creative Professional
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
Don’t cheap out! Use high quality optical fiber (for ADAT) and low-capacitance 
electrical cables (for S/PDIF) when transferring digital I/O to avoid data corruption. 
It’s also a good idea to keep digital cabling as short as possible (1.5 meters for plastic 
light pipes; 5 meters for high quality glass fiber light pipes).
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 pream-
plifier 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.
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, you’ll know if 
you own one. Most microphones are either of dynamic or condenser type and these are 
not harmed by phantom power.
Appearance Settings in Windows
Adjusting the “Performance Options” in Windows will improve the screen appearance 
when moving the mixer around on the screen.
To Improve the Appearance Settings:
1.Open the Windows Control Panel. (Start, Settings, Control Panel).
2.Select System. Select the Advanced Settings tab.
3.Under Visual Effects, select Adjust for Best Performance. Click OK.
1 
(grd)
3
2
+48V 
						

							7 - Appendix
Technical Specifications
E-MU 1820M/1820/1212M PCI Digital Audio System 127
Technical Specifications
Specifications: 1820M System 
GENERAL
Sample Rates44.1 kHz. 48 kHz, 88.2 kHz, 96 kHz, 176.4 kHz, 192 kHz from 
internal crystal. Externally supplied clock from S/PDIF, ADAT 
(or word clock with optional Sync Card)
Bit Depth16 or 24-bits
Hardware DSP100MIPs custom audio DSP. PCI Bus-Mastering DMA subsystem 
reduces CPU usage. Zero-latency direct hardware monitoring 
with effects. 1394 Firewire Core - Texas Instruments
Converters & OpAmpsADC - AK5394 (AKM)
DAC - CS4398 (Cirrus Logic)
OpAmp - NJM2068M (JRC)
WDM Drivers8 channels — operational at 44.1kHz, 48kHz, 88.2kHz, 96kHz, 
176.4kHz & 192kHz
AudioDockM Power Use1.25A @ +12V       15W.
ANALOG LINE INPUTS
TypeServo-balanced, DC-coupled, low-noise input circuitry
Level (software selectable)Professional: +4 dBu nominal, 20 dBu maximum (balanced)
Consumer: -10 dBV nominal, 6 dBV maximum (unbalanced)
Frequency Response+/- .05dB, 20 Hz - 20 kHz
THD + N -110 dB (.0003%) 1kHz at -1 dBFS
SNR120 dB (A-weighted)
Dynamic Range120 dB (A-weighted)
Channel Crosstalk< -115 dB, (1 kHz signal at -1 dBFS)
Common-mode Rejection> 40 dB at 60Hz
Input Impedance10K ohm
ANALOG LINE OUTPUTS
TypeBalanced, low-noise, 2-pole low-pass differential filter
Level (software selectable)Professional: +4dBu nominal, 20dBu (balanced)
Consumer: -10dBV nominal, 6dBV maximum (unbalanced)
Frequency Response+0.0/-0.35 dB, 20 Hz - 20 kHz
THD + N-105 dB (.0006%) 1kHz signal at -1dBFS
SNR120 dB (A-weighted)
Dynamic Range120 dB (A-weighted)
Stereo Crosstalk< -120 dB, 1kHz
Output Impedance560 ohms 
						

							7 - Appendix
Technical Specifications
128Creative Professional
MIC PREAMP/LINE INPUT
TypeTFPro™ combination microphone preamp and line input
Frequency Response+0.8/-0.1 dB, 20 Hz - 20kHz
Stereo Crosstalk< 120 dB, 1kHz
LINE INPUT
Gain Range:-12 to +28 dB
Max Level: -17 dbV (19.2 dBu)
THD+N:-100 dB (.001%), 1 kHz at -1 dBFS
Dynamic Range:107 dB (A-weighted, min. gain)
SNR: 107 dB (A-weighted, min. gain)
Input Impedance: 10K ohm
CMRR: > 40 dB (60Hz)
MICROPHONE PREAMP
Gain Range:-10 to +50 dB
Max Level: -12 dbV (-9.8 dBu)
THD+N:-100 dB (.001%), 1 kHz at -1 dBFS
SNR: 106 dB (A-weighted, min. gain)
Input Impedance: 330 ohm
CMRR: > 80 dB (60Hz)
HEADPHONES
Frequency Response:+0.0/-0.35 dB, 20 Hz - 20 kHz
THD+N: (1 kHz, max. level)33 ohm load: -69 dB (0.035%)
600 ohm load: -94 dB (0.002%)
SNR: 117 dB (A-weighted)
Dynamic Range:117 dB (A-weighted)
Stereo Crosstalk: < -100 dB (1kHz at -1 dBFS, 600 ohm load)
Max Output Power: 500 mW
Output Impedance:22 ohms
Gain Range:85 dB
Specifications: 1820M System  
						

							7 - Appendix
Technical Specifications
E-MU 1820M/1820/1212M PCI Digital Audio System 129
TURNTABLE INPUTRIAA equalized phono input
Frequency Response:+/-0.5 dB, 50 Hz - 20 kHz
THD+N:-76 dB (.015%) (1 kHz, 10 mV RMS unbalanced input)
SNR:90 dB (1kHz, 10 mV RMS unbalanced input)
Stereo Crosstalk:< -80 dB (1kHz at -1 dBFS)
Maximum Level:Professional: 80 mV RMS
Consumer: 20 mV RMS
Input Capacitance:220 pF
Input Impedance:47K ohm
DIGITAL I/O
S/PDIF• 2 in/2 out coaxial (transformer coupled)
• 2 in/3 out optical (software switchable with ADAT)
• AES/EBU or S/PDIF (switchable under software control)
ADAT• 8 channels, 24-bit @ 44.1/48 kHz
• 4 channels, 24-bit @ 96 kHz
• 2 channels, 24-bit @ 192 kHz
Firewire400 IEEE 1394a port (6-pin)
Compatible with DV cameras or HDs
MIDI2 MIDI in, 2 MIDI out
SYNCHRONIZATION
Internal Crystal Sync:44.1 kHz, 48 kHz, 88.2 kHz, 96 kHz, 176.4 kHz, 192 kHz
ADAT, S/PDIF (optical or coaxial)
Word Clock (sync card only) - (75 ohm termination, switchable)
RMS JITTER @ 44.1K 
(Measured via Audio Precision 2)SRSync SourceRMS jitter in picoseconds 
44.1 kHz internal Crystal 596ps 
44.1 kHz Optical Input      795ps
SMPTEConverts to/from longitudinal time code (LTC) to MIDI time 
code (MTC)
Frame Rates24, 25, 30 drop, 30 non-drop frames/second. 
Compatible with 29.97 fps timecode
ModesRegeneration, stripe and conversion modes
Input Level:0.5 - 4V p-p
Output Level:+4 dBu, -10 dBV (software selectable)
Input Impedance:10K ohm
Specifications: 1820M System  
						

							7 - Appendix
Technical Specifications
130Creative Professional
Specifications: 1820 System
GENERAL
Sample Rates44.1 kHz. 48 kHz, 88.2 kHz, 96 kHz, 176.4 kHz, 192 kHz 
from internal crystal. Externally supplied clock from S/PDIF, 
ADAT (or word clock with optional Sync Card)
Bit Depth16 or 24-bits
Hardware DSP100MIPs custom audio DSP. 
PCI Bus-Mastering DMA subsystem reduces CPU usage.
Zero-latency direct hardware monitoring with effects
1394 Firewire Core - Texas Instruments
Converters & OpAmpsADC - PCM1804 (TI/Burr-Brown)
DAC - CS4392 (Cirrus Logic)
OpAmp - NJM2068M (JRC)
AudioDock Power Use1.1A @ +12V       13W.
ANALOG LINE INPUTS
TypeServo-balanced, DC-coupled, low-noise input circuitry
Level (software selectable)Professional: +4 dBu nominal, 20 dBu maximum (balanced)
Consumer: -10 dBV nominal, 6 dBV maximum (unbalanced)
Frequency Response+0.0/-0.2 dB, 20 Hz - 20 kHz
THD + N -102 dB (.0008%) 1kHz at -1 dBFS
SNR111 dB (A-weighted)
Dynamic Range111 dB (A-weighted)
Channel Crosstalk< -115 dB, (1 kHz signal at -1 dBFS)
Common-mode Rejection> 40 dB at 60Hz
Input Impedance10K ohm
ANALOG LINE OUTPUTS
TypeBalanced, low-noise, 2-pole low-pass differential filter
Level (software selectable)Professional: +4dBu nominal, 20dBu (balanced)
Consumer: -10dBV nominal, 6dBV maximum (unbalanced)
Frequency Response+0.0/-0.8 dB, 20 Hz - 20 kHz
THD + N-98 dB (.0006%) 1kHz signal at -1dBFS
SNR112 dB (A-weighted)
Dynamic Range112 dB (A-weighted)
Stereo Crosstalk< -120 dB, 1kHz
Output Impedance560 ohms