ATTENTION! Do not open chassis – risk of electric shock
The unit has unisolated live parts inside. No user serviceable parts inside.
Refer service to qualified service personnel.
Mains
• The device must be earthed – never use it without proper grounding
• Do not use defective power cords
• Operation of the device is limited to the manual
• Use same type of fuse only
To reduce the risk of fire or electric shock do not expose this device to rain or
moisture. Prevent moisture and water from entering the device. Never leave
a pot with liquid on top of the device. Do not use this product near water, i. e.
swimming pool, bathtub or wet basement. Danger of condensation inside –
don't turn on before the device has reached room temperature.
Installation
Surface may become hot during operation – ensure sufficient ventilation.
Avoid direct sun light and do not place it near other sources of heat, like radiators or stoves. When mounting in a rack, leave some space between this
device and others for ventilation.
Unauthorized servicing/repair voids warranty. Only use accessories
specified by the manufacturer.
Read the manual completely. It includes all information necessary
to use and operate this device.
The OctaMic II's innovative concept allows for amplification and digitization of ALL analog signal
sources. Be it high-level stage signals, typical studio signals, lower level and high-impedance
instruments, or dynamic, condenser or ribbon microphones: OctaMic II understands them all –
in a way that is simply thrilling.
When developing the OctaMic II we used all our experience, and also the experience of our
customers, to create a unique, excellent and high-quality unit. As successor of the OctaMic D
the device offers the approved quality of its predecessor, adding lots of improvements in detail.
2. Package Contents
Please check that your OctaMic II package contains each of the following:
• OctaMic II
• Power cord
• Manual
• 1 optical cable (TOSLINK), 2 m
3. Brief Description and Characteristics
• 8 balanced XLR/TRS mic/line inputs
• 54 dB gain range
• Analog input level from –40 dBu up to +21 dBu
• Large frequency range (200 kHz) with special EMI input filtering
• Input impedance: XLR 2 kOhm, TRS 5 kOhm
• Signal to noise ratio (SNR): 129 dB EIN @150 Ohm
• THD: < 0.0005 % @ 30 dB Gain
• Channel seperation: > 110 dB
• Frequency response –0.5 dB: 5 Hz - 200 kHz
• Line Out: 1/4" TRS (6.3 mm stereo jack), servo-balanced
• Maximum output level: +21 dBu
• Output impedance: 75 Ohm
• Output level switchable Hi Gain / +4 dBu / -10 dBV
Each individual OctaMic II undergoes comprehensive quality control and a complete test at IMM
before shipping. The usage of high grade components allow us to offer a full two year warranty.
We accept a copy of the sales receipt as valid warranty legitimation.
If you suspect that your product is faulty, please contact your local retailer. The warranty does
not cover damage caused by improper installation or maltreatment - replacement or repair in
such cases can only be carried out at the owner’s expense.
Audio AG does not accept claims for damages of any kind, especially consequential damage.
Liability is limited to the value of the OctaMic II. The general terms of business drawn up by
Audio AG apply at all times.
6. Appendix
RME news and further information can be found on our website:
http://www.rme-audio.com
Distributor:
IMM Elektronik GmbH, Leipziger Strasse 32, D-09648 Mittweida
Trademarks
All trademarks and registered trademarks belong to their respective owners. RME, SyncAlign,
Hammerfall, DIGICheck, SyncCheck and ZLM are registered trademarks of RME Intelligent
Audio Solutions. TotalGain, SteadyClock, OctaMic II and Intelligent Clock Control (ICC) are
trademarks of RME Intelligent Audio Solutions. Alesis and ADAT are registered trademarks of
Alesis Corp. ADAT optical is a trademark of Alesis Corp. S/MUX is copyright Sonorus.
rectness can be given. RME cannot be held responsible for any misleading or incorrect information provided throughout this manual. Lending or copying any part or the complete manual or its
contents as well as the software belonging to it is only possible with the written permission from
RME. RME reserves the right to change specifications at any time without notice.
This device has been tested and found to comply with the limits of the European Council Directive on the approximation of the laws of the member states relating to electromagnetic compatibility according to RL89/336/EWG and RL73/23/EWG.
FCC
This equipment has been tested and found to comply with the limits for a Class B digital device,
pursuant to Part 15 of the FCC Rules. These limits are designed to provide reasonable protection against harmful interference in a residential installation. This equipment generates, uses,
and can radiate radio frequency energy and, if not installed and used in accordance with the
instructions, may cause harmful interference to radio communications. However, there is no
guarantee that interference will not occur in a particular installation. If this equipment does
cause harmful interference to radio or television reception, which can be determined by turning
the equipment off and on, the user is encouraged to try to correct the interference by one or
more of the following measures:
- Reorient or relocate the receiving antenna.
- Increase the separation between the equipment and receiver.
- Connect the equipment into an outlet on a circuit different from that to which the receiver is
connected.
- Consult the dealer or an experienced radio/TV technician for help.
RoHS
This product has been soldered lead-free and fullfills the requirements of the RoHS directive.
ISO 9001
This product has been manufactured under ISO 9001 quality management. The manufacturer,
IMM Elektronik GmbH, is also certified for ISO 14001 (Environment) and ISO 13485 (medical
devices).
Note on Disposal
According to the guide line RL2002/96/EG (WEEE – Directive on Waste
Electrical and Electronic Equipment), valid for all european countries, this
product has to be recycled at the end of its lifetime.
In case a disposal of electronic waste is not possible, the recycling can
also be done by IMM Elektronik GmbH, the manufacturer of the OctaMic II.
For this the device has to be sent free to the door to:
IMM Elektronik GmbH
Leipziger Straße 32
D-09648 Mittweida
Germany
Shipments not prepaid will be rejected and returned on the original sender's costs.
+48V (LED) lights up when phantom power is active.
The CLIP LED has been designed to act like the OVR LEDs of
the ADI-8 series. It lights up 2 dB before the chosen reference
level plus a headroom of 9 dB. At Hi Gain the LED lights up at
+17 dBu output level, selecting +4 dBu it lights up at +11 dBu.
SIG (Signal) indicates the presence of an input signal. The
LED has a detection range of more than 50 dB using multiple
brightness states. With this, SIG acts as useful level control,
helping to set GAIN correctly.
7.2 Controls
GAIN allows for a stepless adjustment of the amplification between +6 dB and +60 dB.
+48V (switch) activates phantom power. Phantom power should only be activated when using
condensor microphones which require such a power supply, and only on the specific channel.
Connecting and disconnecting microphones while phantom power is active causes a high
voltage surge, which can destroy the microphone input stage! Switch phantom power off
before connecting/disconnecting any external device.
The OctaMic II turns on the phantom power smoothly during one second, from 0 to 48 Volts.
This technique is advantageous for the connected microphone as well as the OctaMic II. The
phantom power of the OctaMic II is short-circuit proof. With a maximum load on all eight channels the internal voltage does not drop below 47 Volts.
LO CUT activates a high-pass (bass removal) with 18 dB per octave, at a cut-off-frequency of
80 Hz. This filter can remove rumble, subsonic-noise and other low frequency noises. The LO
CUT of the OctaMic II shows a slow roll-off in the frequency response, without any resonance
rise, and provides low THD figures. At a low 20 Hz, where the fundamental signal is already
attenuated by 34 dB, the distortion is around 0.13 %.
PHASE changes the polarity (180°). Phase cancellations and sound changes can be caused by
using multiple microphones at different places, or wrongly soldered cables. In such cases
PHASE can eliminate the error by adding an additional phase inversion.
Clip Hold is activated by pressing the key for two seconds. As soon
as an overload is detected, the corresponding Clip LED begins to
flash once per second. With this, a momentary overload stays visible
for a longer time. Pressing the key once resets the Clip display.
Pressing the key again for two seconds deactivates the Clip Hold
mode.
Hi Gain / +4 dBu / -10 dBV: Defines the reference level of the Line
Level Outputs, equalling a full scale level of the AD-converters. See
chapter 10.2.
MICROPHONE / LINE INPUTS: 8 8 Neutrik XLR / TRS combo jacks. Thanks to the servo bal-
anced designs and a high maximum input level (maximum input level XLR +14 dBu, TRS +21
dBu) the inputs are universally applicable.
LINE LEVEL OUTPUTS: 8 TRS (stereo) jacks. The electronic output stage is built in a servo
balanced design, handling monaural (unbalanced) and stereo jacks (balanced) correctly.
IEC receptacle for power connection. The specially developed, internal hi-performance switch
mode power supply lets the OctaMic II operate in the range of 100V to 240V AC. It is shortcircuit-proof, has an integrated line-filter, is fully regulated against voltage fluctuations, and suppresses mains interference.
8.2 DIP Switches
The DIP switches are used to configure
the OctaMic II's digital section. The following diagram, also printed on the back of
the unit, shows the meaning of all possible switch positions.
DIP Switch Function
1 External synchronization source AES (D-sub) or word clock (BNC)
2 Clock internal (Master) or external (Slave)
3 Internal Clock 44.1 kHz or 48 kHz
4 Activates Double Speed Mode*
5 Activates Quad Speed Mode*
6 AES output state Professional or Consumer
*Note on DIP switch 4/5:
At internal clock, the switches DS and QS multiply the value set with switch 3 by a factor of 2 or
4. So if switch 3 is set to 48 kHz, switch 4 will turn it into 96 kHz, switch 5 turns it into 192 kHz.
At external clock switch 3 is of no meaning, because the unit is synchronized to the incoming
clock. However, switch 4 and 5 pre-define the frequency range to Single Speed, Double Speed
or Quad Speed. For example if the OctaMic shall operate at 176.4 or 192 kHz, switch 5 has to
be set to the lower position. The OctaMic will now generate an output signal in the Quad speed
range (176.4 or 192 kHz), even with a word clock input signal of only 44.1 kHz, or an AES input
signal signal of 96 kHz.
With external synchronization active (switch 2 down) and both switches 4 and 5 in lower position, the function Follow Clock (FC) is active. In this case the OctaMic II follows the input clock
1:1. A further configuration of the current sample rate range (Single, Double or Quad Speed) is
not required.
The digital inputs of the OctaMic II are used for external synchronization only. In case the clock
shall not be generated internally (operation mode Master), an external synchronization (operation mode Slave) is available via word clock or AES (SPDIF).
The SteadyClock technology of the OctaMic II guarantess exceptional performance in all clock
modes. Thanks to a highly efficient jitter suppression, the AD-conversion always operates on
highest sonic level, being completely independent from the quality of the incoming clock signal.
When the current word clock source fails, the last valid sample rate will be held automatically.
9.1 Wordclock - BNC
The word clock input is activated by DIP switches 1 and 2. Both switches must be set to their
lower position.
Thanks to RME's Signal Adaptation Circuit, the word clock input still works correctly even with
heavily mis-shaped, dc-prone, too small or overshoot-prone signals. Thanks to automatic signal
centering, 300 mV (0.3V) input level are sufficient in principle. An additional hysteresis reduces
sensitivity to 1.0 V, so that over- and undershoots and high frequency disturbances don't cause
a wrong trigger.
The word clock input is shipped as high impedance type (not terminated). A push switch allows
to activate internal termination (75 Ohms). The switch is found beside the BNC socket. Use a
small pencil or similar and carefully push the blue switch so that it snaps into its lock position.
Another push will release it again and de-activate the termination.
Due to the outstanding clock control a synchronization of the output signal to the input signal is
not only possible at identical sample rates, but also at half, quarter, double and quad sample
rates.
Example 1: DIP switch 3/4/5 in upper position results in a sample rate of 44.1 kHz. The external
synchronization source (word clock or AES) can now be 44.1 kHz, 88.2 kHz or 176.4 kHz.
Example 2: DIP switch 3/5 in lower position results in a sample rate of 192 kHz. The external
synchronization (word clock or AES) source can now be 48 kHz, 96 kHz or 192 kHz.
9.2 AES – D-Sub
Using the D-sub connector, an AES, AES/EBU or SPDIF signal can be used for synchronization. Set DIP switch 1 to the upper position and DIP switch 2 to the lower position.
The OctaMic II's synchronization input uses AES 1 (see chapter 11.1). The input is transformerbalanced and ground-free.
Thanks to a highly sensitive input stage, a
SPDIF signal can also be fed by using a
simple cable adapter phono/XLR. To
achieve this, pins 2 and 3 of an XLR plug
are being connected to the two contacts of a
Phono/RCA plug. The ground shield of the
cable is only connected to pin 1 of the XLR
plug.
With AES, a synchronization is possible not only at identical sample rates, but also at half, quarter, double and quad sample rates.
The OctaMic has 8 balanced Mic and Line inputs via 1/4" TRS (stereo) and XLR combo jacks
on the rear panel. The electronic input stage is built in a servo balanced design which handles
unbalanced and balanced signals correctly, automatically adjusting the level reference.
XLR
The pin assignment follows international standards. With XLR, pin 2 is + or hot, pin 3 is – or
cold, pin 1 is ground. Pin 1 is connected to the chassis directly at the socket (AES48).
When using unbalanced cables be sure to connect pin 3 (-) to 1 (ground). Otherwise noise
may occur, caused by the unconnected negative input of the balanced input.
The OctaMic II offers an adjustable amplification from +6 dB up to +60 dB. This equals a sensitivity of +14 dBu down to –40 dBu, referenced to full scale of the AD-converter.
The soft switching, hi-current phantom power (48 Volt) provides a professional handling of condenser microphones. The usage of a hi-end integrated circuit (That 1510) guarantees outstanding sound quality, lowest THD, and maximum Signal to Noise ratio in any gain setting.
The OctaMic II's 'overall' amplification from analog input to analog output depends on the analog output reference setting. The EIN is not affected by this setting, since the S/N ratio scales
1:1 with the output amplification.
The input impedance is 2 kOhm.
TRS
The pin assignment follows international standards. With TRS, tip is + or hot, ring is – or cold.
When using unbalanced cables with stereo TRS jacks, the 'ring' contact of the cable's jack
should be connected to pin 1 (ground). Otherwise noise may occur, caused by the unconnected negative input of the balanced input.
The TRS jacks have a fixed level attenuation of 7 dB. Based on the adjustable amplification
from +6 dB up to +60 dB, the sensitivity is +21 dBu down to –33 dBu, referenced to full scale of
the AD-converter. Therefore the TRS inputs are true full level Line inputs, and the unit casn also
be used as Line amplifier.
The TRS jacks are free of phantom power. The unbalanced input impedance is 5 kOhm.
The 8 short circuit protected, low impedance and servo balanced line outputs are available as
(stereo) 1/4" TRS jacks. The electronic output stage is built in a servo balanced design which
handles monaural and stereo jacks correctly.
The pinout follows international standards. Tip + or hot, ring – or cold.
To maintain an optimum level for devices connected to the analog outputs, the OctaMic in-
cludes a switch which allows to change the reference level of all 8 outputs simultaneously.
The OctaMic II can generate a maximum level of +21 dBu without distortion. However, the CLIP
LED has been designed to act like the OVR LEDs of the ADI-8 series. It lights up 2 dB before
the reference level selected on the back of the unit, plus a headroom of 9 dB, is reached. At Hi
Gain the LED lights up at +17 dBu output level, selecting +4 dBu it lights up at +11 dBu, selecting –10 dBV it lights up at 0 dBV.
Setting Reference Clip LED True Clip ADC Level
Hi Gain +19 dBu +17 dBu +21 dBu -2 dBFS
+4 dBu +13 dBu +11 dBu +15 dBu -2 dBFS
-10 dBV +2 dBV 0 dBV +4 dBV -2 dBFS
This also means that the CLIP LED lights up 4 dB before the OctaMic II actually reaches the
maximum level. Such an additional headroom is considered to be useful in real world operation.
The chosen reference level has no meaning for the digital outputs. The AD-conversion is designed for a level of –2 dBFS as soon as the Clip LED lights up.
Selecting +4 dBu the output signal is attenuated by 6 dB, so for the same output level the amplification has to be increased via GAIN. With this trick the OctaMic reaches the maximum signal
to noise ratio on +4 dBu based inputs (like our ADI-8 series), because microphone preamps
have better EIN values at higher amplification. In case of an extreme recording situation, where
the gain of the OctaMic is no longer sufficient, selecting Hi Gain will again provide the highest
amplification possible.
The same is true and even more efficient at –10 dBV. In this case the output level is reduced by
around 14 dB – the same happens to the basic noise of the unit!
The four AES/EBU outputs are provided on the rear of the OctaMic II via a 25 pin D-sub connector with Tascam pinout (also used by Digidesign). A digital breakout cable will provide 4
male (and 4 female) XLR connectors. Every output is transformer-balanced, ground-free and
compatible to all devices with AES/EBU ports.
Besides the audio data, digital signals in SPDIF or AES/EBU format contain a channel status
coding, which is being used for transmitting further information. The output signal coding of the
OctaMic II has been implemented according to AES3-1992 Amendment 4:
• 32* / 44.1 / 48 / 64* / 88.2 / 96 / 176.4 / 192 kHz according to the current sample rate
• Audio use
• No Copyright, Copy permitted
• Format Professional
• Category General, Generation not indicated
• 2-Channel, No Emphasis
• Aux bits audio use, 24 bit
• Origin: 8MIC
* This status is set automatically in external synchronization mode as soon as the sample rate is
recognized.
The format is set by DIP switch 6 to Professional or Consumer. If AES PRO is chosen, the output level is almost 5V. If CON (Consumer) is chosen, the output signal will have a channel
status compatible to SPDIF, and the output level will be reduced to 2V.
Connecting devices with coaxial SPDIF
ports to the OctaMic II's outputs is accomplished by simple cable adapters
XLR/RCA.. To achieve this, pins 2 and 3 of
an XLR plug are being connected to the two
contacts of a Phono/RCA plug. The ground
shield of the cable is only connected to pin 1
of the XLR plug.
Note that most consumer HiFi equipment with phono SPDIF inputs will only accept signals
with Channel Status ‘Consumer’.
The OctaMic II supports Single Wire only, in the range of 32 kHz up to 192 kHz: a total of 8
chanels, 2 channels per AES wire. The effective sample frequency equals the clock on the AES
wire. In case a conversion from/to Single, Double and Quad Wire is required, the RME ADI-192
DD, an 8-channel universal sample rate and format converter, is highly recommended.
Pinout of the D-sub connector, Outputs
Signal Out
1/2+
D-Sub 18 6 4 17 15 3 1 14
GND is connected to pins 2, 5, 8, 11, 16, 19, 22, 25. Pin 13 is not connected.
The input AES 1 (channel 1/2) found on the D-sub connector can not be used for audio with the
OctaMic II, but as clock source. The input is transformer-balanced and ground-free. Thanks to a
highly sensitive input stage, a SPDIF signal can also be fed by using a simple cable adapter
phono/XLR. (see above).
Pinout of the D-sub connector, Inputs
Signal In
1/2+
In
1/2-
In
3/4+
In
3/4-
In
5/6+
In
5/6-
In
7/8+
In
7/8-
D-sub 24 12 10 23 21 9 7 20
GND is connected to pins 2, 5, 8, 11, 16, 19, 22, 25. Pin 13 is not connected.
11.2 ADAT Optical
The OctaMic II provides two digital outputs in ADAT optical format. In Single Speed mode both
outputs carry the same audio data. With this it is possible to distribute the output signal to two
devices (2 x ADAT splitter).
As the ADAT optical signal is physically specified up to 48 kHz only, the OctaMic II automatically activates Sample Split mode (S/MUX) at 88.2 and 96 kHz, distributing the data of one
channel to two output channels. The internal frequency stays at 44.1/48 kHz. Therefore the
sample clock at the ADAT outputs is only half the frequency of the AES outputs. As interesting
as this is – you don't need to think about it. 96 kHz capable ADAT hardware, like all current
RME digital interfaces, re-combine the data automatically. The user (and the DAW software)
does not see any split data, but just single channels at the expected double sample rate.
Up to 96 kHz (Double Speed), the ADAT outputs can be used in parallel to the AES outputs. In
Quad Speed mode (128 kHz up to 192 kHz), the ADAT outputs are operated synchronously at
Single Speed sample clock, but do not provide any audio data.
The ADAT optical outputs of the ADC Module are fully compatible to all ADAT optical inputs. A
usual TOSLINK cable is sufficient for connection.
ADAT Main
Interface for the first or only device receiving an ADAT signal from the OctaMic. Carries the
channels 1 to 8. When sending a Double Speed signal, this port carries the channels 1 to 4. In
Quad Speed mode ADAT MAIN carries an empty but synchronous ADAT signal.
ADAT AUX
Copy of the data at the MAIN output. When sending a Double Speed signal, this port carries the
channels 5 to 8. In Quad Speed mode ADAT AUX carries an empty but synchronous ADAT
signal.
In the analog domain one can connect any device to another device, a synchronization is not
necessary. Digital audio is different. It uses a clock, the sample frequency. The signal can only
be processed and transmitted when all participating devices share the same clock. If not, the
signal will suffer from wrong samples, distortion, crackle sounds and drop outs.
AES/EBU, SPDIF, ADAT and MADI are self-clocking, an additional word clock connection in
principle isn't necessary. But when using more than one device simultaneously problems are
likely to happen. For example any self-clocking will not work in a loop cabling, when there is no
'master' (main clock) inside the loop. Additionally the clock of all participating devices has to be
synchronous. This is often impossible with devices limited to playback, for example CD players,
as these have no SPDIF input, thus can't use the self clocking technique as clock reference.
In a digital studio synchronisation is maintained by connecting all devices to a central sync
source. For example the mixing desk works as master and sends a reference signal, the word
clock, to all other devices. Of course this will only work as long as all other devices are
equipped with a word clock or sync input, thus being able to work as slave (some professional
CD players indeed have a word clock input). Then all devices get the same clock and will work
in every possible combination with each other.
Remember that a digital system can only have one master! If the OctaMic II uses its internal
clock, all other devices must be set to ‘Slave’ mode.
But word clock is not only the 'great problem solver', it also has some disadvantages. The word
clock is based on a fraction of the really needed clock. For example SPDIF: 44.1 kHz word
clock (a simple square wave signal) has to be multiplied by 256 inside the device using a special PLL (to about 11.2 MHz). This signal then replaces the one from the quartz crystal. Big
disadvantage: because of the high multiplication factor the reconstructed clock will have great
deviations called jitter. The jitter of a word clock is much higher as when using a quartz based
clock.
The end of these problems should have been the so called Superclock, which uses 256 times
the word clock frequency. This equals the internal quartz frequency, so no PLL for multiplying is
needed and the clock can be used directly. But reality was different, the Superclock proved to
be much more critical than word clock. A square wave signal of 11 MHz distributed to several
devices - this simply means to fight with high frequency technology. Reflections, cable quality,
capacitive loads - at 44.1 kHz these factors may be ignored, at 11 MHz they are the end of the
clock network. Additionally it was found that a PLL not only generates jitter, but also also rejects
disturbances. The slow PLL works like a filter for induced and modulated frequencies above
several kHz. As the Superclock is used without any filtering such a kind of jitter and noise suppression is missing. In the end Superclock did not become a commonly accepted standard.
The actual end of these problems is offered by the SteadyClock technology of the OctaMic II.
Combining the advantages of modern and fastest digital technology with analog filter techniques, re-gaining a low jitter clock signal of 22 MHz from a slow word clock of 44.1 kHz is no
problem anymore. Additionally, jitter on the input signal is highly rejected, so that even in real
world usage the re-gained clock signal is of highest quality.
Word clock signals are usually distributed in the form of a network, split with BNC T-adapters
and terminated with resistors. We recommend using off-the-shelf BNC cables to connect all
devices, as this type of cable is used for most computer networks. Actually you will find all the
necessary components (T-adapters, terminators, cables) in most electronics and computer
stores. The latter usually carries 50 Ohm components. The 75 Ohm components used for word
clock are part of video techology (RG59).
Ideally, the word clock signal is a 5 Volt square wave with the frequency of the sample rate, of
which the harmonics go up to far above 500 kHz. To avoid voltage loss and reflections, both the
cable itself and the terminating resistor at the end of the chain should have an impedance of 75
Ohm. If the voltage is too low, synchronization will fail. High frequency reflection effects can
cause both jitter and sync failure.
Unfortunately there are still many devices on the market, even newer digital mixing consoles,
which are supplied with a word clock output that can only be called unsatisfactory. If the output
breaks down to 3 Volts when terminating with 75 Ohms, you have to take into account that a
device, of which the input only works from 2.8 Volts and above, does not function correctly already after 3 meter cable length. So it is not astonishing that because of the higher voltage,
word clock networks are in some cases more stable and reliable if cables are not terminated at
all.
Ideally all outputs of word clock delivering devices are designed as low impedance types, but all
word clock inputs as high impedance types, in order to not weaken the signal on the chain. But
there are also negative examples, when the 75 Ohms are built into the device and cannot be
switched off. In this case the network load is often 2 x 75 Ohms, and the user is forced to buy a
special word clock distributor. Note that such a device is generally recommended for larger
studios.
The OctaMic II's word clock input can be high-impedance or terminated internally, ensuring
maximum flexibility. If termination is necessary (e.g. because the OctaMic II is the last device in
the chain), push the switch at the back (see chapter 9.1).
In case the OctaMic II resides within a chain of devices receiving word clock, plug a T-adapter
into its BNC input jack, and the cable supplying the word clock signal to one end of the adapter.
Connect the free end to the next device in the chain via a further BNC cable. The last device in
the chain should be terminated using another T-adapter and a 75 Ohm resistor (available as
short BNC plug). Of course devices with internal termination do not need T-adaptor and terminator plug.
The pinout of the 25-pin D-sub connector uses the widely spread Tascam scheme, which is
also used by Digidesign. Out of the four AES inputs only the first one is used by the OctaMic
(Sync).
Tascam / Digidesign:
Signal In
1/2+
D-Sub 24 12 10 23 21 9 7 20
Signal Out
1/2+
D-Sub 18 6 4 17 15 3 1 14
GND is connected to pins 2, 5, 8, 11, 16, 19, 22, 25. Pin 13 is not connected.
The Yamaha pinout is quite popular as well. When building a D-sub to D-sub adapter or con-
nection cable, please make sure that the connectors are clearly labeled with Tascam and Ya-maha. The cable can only be used when the Tascam side is connected to a Tascam connector,
and the Yamaha side is connected to a Yamaha connector.
Yamaha:
Signal In
1/2+
D-Sub 1 14 2 15 3 16 4 17
Signal Out
1/2+
D-Sub 5 18 6 19 7 20 8 21
GND is connected to pins 9, 10, 11, 12, 13, 22, 23, 24, 25.
The same is true for a direct adapter cable Tascam D-sub to Euphonix D-sub.
Euphonix:
Signal In
1/2+
D-Sub 15 2 4 16 18 5 7 19
Signal Out
1/2+
D-Sub 21 8 10 22 24 11 13 25
GND is connected to pins 3, 6, 9, 12, 14, 17, 20, 23. Pin 1 is not connected.
XLR sockets analog input and output
The XLR connectors of the analog inputs and outputs are wired according to international stan-
dards:
1 = GND (shield)
2 = + (hot)
3 = - (cold)
The servo balanced input and output circuitry allows to use unbalanced connections with no
loss in level. For this to work, pins 3 (-) and 1 (GND) have to be connected inside the XLR connector.
TRS jacks analog input and output
The stereo ¼" TRS jacks of the analog inputs and outputs are wired according to international
standards:
Tip = + (hot)
Ring = – (cold)
Sleeve = GND
The servo balanced input and otput circuitry allows to use monaural TS jacks (unbalanced) with
no loss in level. This is the same as when using a TRS-jack with ring connected to ground.
Sample rate range originally used in Digital Audio. Typical applications are 32 kHz (digital radio
broadcast), 44.1 kHz (CD), and 48 kHz (DAT).
Double Speed
Doubles the original sample rate range, in order to achieve higher audio quality and improved
audio processing. 64 kHz is practically never used, 88.2 kHz is quite rare in spite of certain advantages. 96 kHz is a common format. Sometimes called Double Fast.
Quad Speed
Controversially discussed way of ensuring hi-end audio quality and processing by quadrupling
the sample frequency. 128 kHz is non-existant, 176.4 kHz is rare, if at all then 192 kHz is used,
e.g. for DVD Audio.
Single Wire
Standard audio data transfer, where the audio signal's sample rate is equal to the rate of the
digital signal. Used from 32 to 192 kHz. Sometimes called Single Wide.
Double Wire
Before 1998 there were no receiver/transmitter circuits available that could receive or transmit
more than 48 kHz. Higher sample rates were transferred by splitting odd and even bits across
the L/R channels of a single AES connection. This provides for twice the data rate, and hence
twice the sample rate. A stereo signal subsequently requires two AES/EBU ports.
The Double Wire method is an industry standard today, however it has a number of different
names, like Dual AES, Double Wide, Dual Line and Wide Wire. The AES3 specification uses
the uncommon term Single channel double sampling frequency mode. When used with the
ADAT format, the term S/MUX is commonly used.
Double Wire not only works with Single Speed signals, but also with Double Speed. As an example, Pro Tools HD, whose AES receiver/transmitter only work up to 96 kHz, uses Double
Wire to transmit 192 kHz. Four channels of 96 kHz turn into two channels of 192 kHz.
Quad Wire
Similar to Double Wire, with samples of one channel spread across four channels. This way
single speed devices can transmit up to 192 kHz, but need two AES/EBU ports to transmit one
channel. Also called Quad AES.
S/MUX
Since the ADAT hardware interface is limited to Single Speed, the Double Wire method is used
for sample rates up to 96 kHz, but usually referred to as S/MUX (Sample Multiplexing). An
ADAT port supports four channels this way.
S/MUX4
The Quad Wire method allows to transmit two channels at up to 192 kHz via ADAT. The
method is referred to as S/MUX4.
: All conversions of the described methods are lossless. The existing samples are just
When activating the Double Speed mode the OctaMic II operates at double sample rate. The
internal clock 44.1 kHz turns to 88.2 kHz, 48 kHz to 96 kHz. The internal resolution is still 24 bit.
Sample rates above 48 kHz were not always taken for granted, and are still not widely used
because of the CD format (44.1 kHz) dominating everything. Before 1998 there were no receiver/transmitter circuits available that could receive or transmit more than 48 kHz. Therefore a
work-around was used: instead of two channels, one AES line only carries one channel, whose
odd and even samples are being distributed to the former left and right channels. By this, you
get the double amount of data, i. e. also double sample rate. Of course in order to transmit a
stereo signal two AES/EBU ports are necessary then.
This transmission mode is called Double Wire in the professional studio world, and is also
known as S/MUX (Sample Multiplexing) in connection with the ADAT format.
Not before February 1998, Crystal shipped the first 'single wire' receiver/transmitters that could
also work with double sample rate. It was then possible to transmit two channels of 96 kHz data
via one AES/EBU port.
But Double Wire is still far from being dead. On one hand, there are still many devices which
can't handle more than 48 kHz, e. g. digital tape recorders. But also other common interfaces
like ADAT or TDIF are still using this technique.
Because the ADAT interface does not allow for sampling frequencies above 48 kHz (a limitation
of the interface hardware), the Fireface 800 automatically uses Sample Multiplexing in DS
mode. One channel's data is distributed to two channels according to the following table:
Analog In 1 2 3 4 5 6 7 8
DS Signal
Port
1/2
ADAT1
3/4
ADAT1
5/6
ADAT1
7/8
ADAT1
1/2
ADAT2
3/4
ADAT2
5/6
ADAT2
7/8
ADAT2
As the transmission of double rate signals is done at standard sample rate (Single Speed), the
ADAT outputs still deliver 44.1 kHz or 48 kHz.
14.3 QS – Quad Speed
Due to the small number of available devices that use sample rates up to 192 kHz, but even
more due to a missing real world application (CD...), Quad Speed has had no broad success so
far. An implementation of the ADAT format as double S/MUX (S/MUX4) results in only two
channels per optical output. Therefore the OctaMic II does not support this mode at the ADAT
outputs.
The AES outputs provide 192 kHz as Single Wire only.
The most important electrical properties of 'AES' and 'SPDIF' can be seen in the table below.
AES/EBU is the professional balanced connection using XLR plugs. The standard is being set
by the Audio Engineering Society based on the AES3-1992. For the 'home user', SONY and
Philips have omitted the balanced connection and use either Phono plugs or optical cables
(TOSLINK). The format called S/P-DIF (SONY/Philips Digital Interface) is described by IEC
60958.
Type AES3-1992 IEC 60958
Connection XLR RCA / Optical
Mode Balanced Unbalanced
Impedance 110 Ohm 75 Ohm
Level 0.2 V up to 5 Vpp 0.2 V up to 0.5 Vpp
Clock accuracy not specified
Besides the electrical differences, both formats also have a slightly different setup. The two
formats are compatible in principle, because the audio information is stored in the same place in
the data stream. However, there are blocks of additional information, which are different for both
standards. In the table, the meaning of the first byte (#0) is shown for both formats. The first bit
already determines whether the following bits should be read as Professional or Consumer
information.
Byte Mode Bit 0 1 2 3 4 5 6 7
0 Pro P/C Audio? Emphasis Locked Sample Freq.
0 Con P/C Audio? Copy Emphasis Mode
It becomes obvious that the meaning of the following bits differs quite substantially between the
two formats. If a device like a common DAT recorder only has an SPDIF input, it usually understands only this format. In most cases, it will switch off when being fed Professional-coded data.
The table shows that a Professional-coded signal would lead to malfunctions for copy prohibition and emphasis, if being read as Consumer-coded data.
Nowadays many devices with SPDIF input can handle Professional subcode. Devices with
AES3 input almost always accept Consumer SPDIF (passive cable adapter required).
The SteadyClock technology of the OctaMic II guarantees an excellent performance in all clock
modes. Its highly efficient jitter suppression refreshes and cleans up any clock signal, and provides it as reference clock at the word clock output.
Usually a clock section consists of an analog PLL for external synchronization and several
quartz oscillators for internal synchronisation. SteadyClock requires only one quartz, using a
frequency not equalling digital audio. Latest circuit designs like hi-speed digital synthesizer,
digital PLL, 100 MHz sample rate and analog filtering allow RME to realize a completely newly
developed clock technology, right within the FPGA at lowest costs. The clock's performance
exceeds even professional expectations. Despite its remarkable features, SteadyClock reacts
quite fast compared to other techniques. It locks in fractions of a second to the input signal,
follows even extreme varipitch changes with phase accuracy, and locks directly within a range
of 28 kHz up to 200 kHz.
SteadyClock has originally been developed to gain a stable and clean
clock from the heavily jittery MADI data
signal. The embedded MADI clock
suffers from about 80 ns jitter, caused
by the time resolution of 125 MHz
within the format. Common jitter values
for other devices are 5 ns, while a very
good clock will have less than 2 ns.
The picture to the right shows the
MADI input signal with 80 ns of jitter
(top graph, yellow). Thanks to SteadyClock this signal turns into a clock with
less than 2 ns jitter (lower graph, blue).
Using the input sources of the OctaMic
II, word clock and AES/EBU, you'll
most probably never experience such
high jitter values. But SteadyClock is
not only ready for them, it would handle
them just on the fly.
The screnshot to the right shows an
extremely jittery word clock signal of
about 50 ns jitter (top graph, yellow).
Again SteadyClock provides an extreme clean-up. The filtered clock
shows less than 2 ns jitter (lower
graph, blue).
The cleaned and jitter-freed signal can be used as reference clock for any application, without
any problem. The signal processed by SteadyClock is of course not only used internally, but is
also used to clock the digital outputs ADAT and AES/EBU.