Thank you for choosing the Fireface UC. This unique audio system is capable of transferring
analog and digital audio data directly to a computer from practically any source. The latest Plug
and Play technology guarantees a simple installation, even for the inexperienced user. The
numerous unique features and well thought-out configuration dialog puts the Fireface UC at the
very top of the range of computer-based audio interfaces.
The package contains drivers for Windows XP, XP 64, Vista, Vista 64, Windows 7 /64 and Mac
OS X Intel.
Our high-performance philosophy guarantees maximum system performance by executing as
many functions as possible not in the driver (i.e. the CPU), but directly within the audio hardware.
2. Package Contents
Please check that your Fireface UC package contains each of the following:
• Fireface UC
• Cable USB 2.0, 1.8 m (6 ft)
• MIDI breakout cable
• Power supply
• Manual
• RME Driver CD
• 1 optical cable (TOSLINK), 2 m (6.6 ft)
3. System Requirements
• Windows XP SP or up, Intel Mac OS X (10.5 or up)
The front of the Fireface UC features instrument, microphone and line inputs, a stereo
line/headphone output, a rotary encoder with 7 segment display, and several status and MIDI
LEDs.
The Neutrik combo jacks of the
Mic/Line inputs can be used via
XLR and 1/4" TRS plugs. Both
inputs display overload (CLIP),
signal presence (SIG) and phantom power (48V) via green, red
and yellow LEDs.
Inputs 3/4, INST/LINE, accept both a balanced line signal as well as an unbalanced instrument
signal via 1/4" TRS plug.
The rotary encoder serves to set the input and output levels directly at the unit. This is not only
useful in stand-alone operation, but for example also when setting up the monitor volume.
Pushing the knob changes the encoder from CHANNEL to LEVEL mode and back. Pushing the
knob for more than a second activates either the single channel or stereo setup mode.
The State LEDs (WC, SPDIF, ADAT) indicate a valid input signal separately for each digital
input. Additionally, RME's exclusive SyncCheck indicates if one of these inputs is locked, but
not synchronous to the others, in which case the LED will flash. See also chapter 11.5/19.2,
Clock Modes - Synchronization.
The red HOST LED lights up when the
Fireface UC has been switched on. It
operates as error LED, in case the FireWire
connection hasn't been initialised yet, or has
been interrupted (error, cable not connected
etc.).
The yellow MIDI LEDs indicate MIDI data
received or sent, separately for both inputs
and outputs.
Phones is a low impedance line output of highest quality. It provides a sufficient and undistorted volume even when used with headphones.
The rear panel of the Fireface UC features four analog inputs, six analog outputs, the power
socket, and all digital inputs and outputs.
SPDIF I/O coaxial (RCA): AES/EBU compatible. The Fireface UC accepts the commonly used
digital audio formats, SPDIF as well as AES/EBU.
ADAT I/O (TOSLINK):
The unit automatically
detects SPDIF or ADAT
input signals. The optical
output can operate as
ADAT or SPDIF output,
depending on the current
setting in the Settings
dialog.
Word Clock I/O (BNC): A push switch activates internal termination (75 Ohms). When termination is activated the yellow LED beside the switch lights up.
MIDI I/O: Provides two MIDI inputs and outputs via the included breakout cable.
USB 2.0: USB socket for connection to the computer.
POWER (switch): Turns the Fireface UC on and off.
Socket for power connection. The
included hi-performance switch
mode power supply makes the
Fireface 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.
After the driver installation (chapter 10 / 18) connect the TRS jacks or the XLR jacks with the
analog signal source. The input sensitivity of the rear inputs can be changed in the Settings
dialog (Gain/Level), assuring the highest signal to noise ratio will be achieved. Try to achieve an
optimum input level by adjusting the source itself. Raise the source’s output level until the peak
level meters in TotalMix reach about –3 dB.
The analog line inputs of the Fireface UC can be used with +4 dBu and -10 dBV signals. The
electronic input stage can handle balanced (XLR, TRS jacks) and unbalanced (TS jacks) input
signals correctly.
The front's inputs signal level can be optimized using the Fireface's rotary encoder. A Signal
LED and a Clip LED help to find the correct level adjustment.
The Fireface's digital outputs provide SPDIF (AES/EBU compatible) and ADAT optical signals
at the corresponding ports.
On the analog playback side (the DA side), a coarse adjustment of the analog output level at
the rear jacks is available in the Settings dialog (Gain/Level/Line Out).
The output signal of channels 7/8 is available on the front. Their output level can be set using
the rotary encoder. This output is a very low impedance type, which can also be used to connect headphones.
The Fireface UC remembers all settings, and loads these automatically when switched on. With
this, the Fireface UC can be used stand-alone after setting it up accordingly, replacing lots of
dedicated devices (see chapter 24).
6. Accessories
RME offers several optional components for the Fireface UC:
Part Number Description
Optical cable for SPDIF and ADAT operation:
OK0050 Optical cable, TOSLINK, 0.5 m (1.6 ft)
OK0100 Optical cable, TOSLINK, 1 m (3.3 ft)
OK0200 Optical cable, TOSLINK, 2 m (6.6 ft)
OK0300 Optical cable, TOSLINK, 3 m (9.9 ft)
OK0500 Optical cable, TOSLINK, 5 m (16.4 ft)
OK1000 Optical cable, TOSLINK, 10 m (33 ft)
NTCARDBUS Power supply for Fireface UC. Robust and light-weight switching power sup-
Each individual Fireface UC undergoes comprehensive quality control and a complete test at
IMM before shipping. The usage of high grade components allows 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 Fireface UC. The general terms of business drawn up by
Audio AG apply at all times.
8. Appendix
RME news, driver updates and further product information are available on our website:
http://www.rme-audio.com
Distributor: Audio AG, Am Pfanderling 60, D-85778 Haimhausen, Tel.: (49) 08133 / 91810
Manufacturer:
IMM Elektronik GmbH, Leipziger Strasse 32, D-09648 Mittweida
Trademarks
All trademarks, registered or otherwise, are the property of their respective owners. RME,
DIGICheck and Hammerfall are registered trademarks of RME Intelligent Audio Solutions.
SyncCheck, ZLM, DIGI96, SyncAlign, TMS, TotalMix, SteadyClock and Fireface are trademarks
of RME Intelligent Audio Solutions. Alesis and ADAT are registered trademarks of Alesis Corp.
ADAT optical is a trademark of Alesis Corp. Microsoft, Windows XP and Windows Vista are
registered trademarks or trademarks of Microsoft Corp. Steinberg, Cubase and VST are registered trademarks of Steinberg Media Technologies GmbH. ASIO is a trademark of Steinberg
Media Technologies GmbH.
Although the contents of this User’s Guide have been thoroughly checked for errors, RME can
not guarantee that it is correct throughout. RME does not accept responsibility for any misleading or incorrect information within this guide. Lending or copying any part of the guide or the
RME Driver CD, or any commercial exploitation of these media without express written permission from RME Intelligent Audio Solutions is prohibited. 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 RL2004/108/EG, and European Low Voltage Directive RL2006/95/EG.
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 fulfils 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 Fireface UC.
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.
• Connect the power supply to the Fireface and then to any suitable power outlet. Power-on
the Fireface with the rear power switch.
• Check the correct firmware version by a double click on the rotary encoder button. The
display PC means Windows, the display AP means Mac. The double click automatically
switches between both versions. Note: a change of state is only possible when the unit is
not connected to the computer.
• Connect computer and Fireface using the supplied USB cable.
• Windows detects the new hardware as Fireface UC Win (serial number) and asks for
drivers.
10. Driver and Firmware
10.1 Driver Installation
After the Fireface has been recognized, (see 9. Hardware Installation) the hardware assistant
finds a Fireface UC Win (serial number). Please note the Win. If Mac is displayed the unit has
to be disconnected from the computer and its firmware has to be changed to Win (see above)
prior to installing the drivers.
Insert the RME Driver CD into your CD-ROM drive, and follow further instructions which appear
on your computer screen. The driver files are located in the directory \Fireface UC Win on the
RME Driver CD.
Windows now installs the driver of the Fireface UC and registers it
as a new audio device in the system. After a reboot, the symbols of
mixer and Settings dialog will appear in the task bar. The red Host
error LED extinguishes.
In case the warning messages 'Digital signature not found', 'Do not install driver', 'not certified
driver' or similar come up: simply ignore them and continue with the installation.
In case the Hardware Wizard does not show up automatically after connecting the Fireface,
do not attempt to install the drivers manually! An installation of drivers for non-recognized
hardware will cause a blue screen when booting Windows!
Possible reasons why a Fireface is not found automatically:
• The USB port is not active in the system (check the Device Manager)
• The USB cable is not, or not correctly inserted into the socket
• No power. After switching the Fireface on, at least the 7 segment display has to be lit.
When facing problems with the automatic driver update, the user-driven way of driver installation will work.
Under >Control Panel /System /Device Manager /Sound, Video and Game Controllers /RME Fireface UC/Properties /Driver< you'll find the 'Update Driver' button.
XP: Select 'Install from a list or specific location (advanced)', click 'Next', select 'Don't
search I will choose the driver to install', click 'Next', then 'Have Disk'. Now point to the
driver update's directory.
Vista: Select 'Browse my computer for driver software', then 'Let me pick from a list of
device drivers from my computer', then 'Have Disk'. Now point to the driver update's direc-
tory.
This method also allows for the installation of older drivers than the currently installed ones.
10.3 Deinstalling the Drivers
A deinstallation of the driver files is not necessary – and not supported by Windows anyway.
Thanks to full Plug & Play support, the driver files will not be loaded after the hardware has
been removed. If desired these files can then be deleted manually.
Unfortunately Windows Plug & Play methods do not cover the additional autorun entries of TotalMix, the Settings dialog, and the registration of the ASIO driver. These entries can be removed from the registry by a software deinstallation request. This request can be found (like all
deinstallation entries) in Control Panel, Software. Click on the entry 'RME Fireface UC'.
10.4 Firmware Update
The Flash Update Tool updates the firmware of the Fireface UC to the latest version. It requires
an already installed driver.
Start the program fut_usb.exe. The Flash Update Tool displays the current revision of the Fireface UC's firmware, and whether it needs an update or not. If so, then simply press the 'Update'
button. A progress bar will indicate when the flash process is finished (Verify Ok).
If more than one unit is installed, further units can be flashed by changing to the next tab and
repeating the process.
After the update the Fireface UC needs to be reset. This is done by powering down the Fireface
for a short time. Attention: the Fireface should not be switched off for less than 5 seconds, because Windows completely unloads the driver, which takes some time to finish.
A reboot of the computer is not necessary.
When the update fails (status: failure), the unit's Safety-BIOS will be used from the next boot on,
the unit stays fully functional. The flash process should then be tried again.
In case the automatic activation of the Safety-BIOS fails it can also be activated manually: Push
the rotary encoder button while switching on the unit. This method is also useful to temporarily
deactivate a newer firmware. With pressed button the older version of the firmware will be
loaded (at the time of writing this manual version 82).
Configuration of the Fireface UC is done via its own settings dialog. The panel 'Settings' can be
opened:
• by clicking on the fire symbol in the Task Bar's notification area
The mixer of the Fireface UC (TotalMix) can be opened:
• by clicking on the mixer icon in the Task Bar's notification area
The hardware of the Fireface UC offers a number of helpful, well thought-of practical functions
and options which affect how the card operates - it can be configured to suit many different
requirements. The following is available in the 'Settings' dialog:
• Latency
• Configuration of digital I/Os
• Current sample rate
• Synchronization behaviour
• State of input and output
• Level of analog I/Os
Any changes made in the Settings
dialog are applied immediately confirmation (e.g. by clicking on OK
or exiting the dialog) is not required.
However, settings should not be
changed during playback or record
if it can be avoided, as this can
cause unwanted noises. Also,
please note that even in 'Stop'
mode, several programs keep the
recording and playback devices
open, which means that any new
settings might not be applied immediately.
The status displays at the bottom of
the dialog box give precise
information about the current status
of the system, and the status of all
digital signals.
Input Status indicates for each input (Word, optical, SPDIF coax.) whether there is a valid signal
(Lock, No Lock), or if there is a valid and synchronous signal (Sync). The Clock Mode display
shows the clock reference (Current…).
The string Errors does not refer to buffer errors, but USB transmission errors. More information
can be found in chapter 35.3.
The setting Buffer Size determines the latency between incoming and outgoing ASIO and WDM
data, as well as affecting system stability (see chapter 13/14).
The string Errors does not refer to buffer errors, but USB transmission errors. The display will
be reset on any start of a playback/record. More information can be found in chapter 35.3.
Output Format
Word
The word clock output signal usually equals the current sample rate. Selecting Single Speed
causes the output signal to always stay within the range of 32 kHz to 48 kHz. So at 96 kHz and
192 kHz sample rate, the output word clock is 48 kHz.
Optical
The optical TOSLINK output can operate as ADAT or SPDIF output. The Channel Status is
fixed to Consumer state.
: The optical input detects the incoming format automatically.
Note
SPDIF coax.
The coaxial SPDIF output can have the Channel Status Consumer or Professional. For further
details please refer to chapter 27.2.
Clock Mode
Sample Rate
Sets the currently used sample rate. Offers a central and comfortable way of configuring the
sample rate of all WDM devices to the same value, as since Vista this is no longer supported to
be done by the audio program. However, an ASIO program can still set the sample rate by itself.
At ongoing record/playback the selection is greyed out, so no change is possible.
Clock Source
The unit can be configured to use its own clock (Internal = Master), or one of the input signals
(Word, Optical, SPDIF coax., LTC = Slave). If the selected source isn't available, the unit will
change to the next available one (AutoSync). If none is available then the internal clock is used.
The current clock source is displayed to the right.
Pitch
More information on Pitch is available in chapter 11.2.
Input Status
Indicates for each input (Word, optical, SPDIF coax.) whether there is a valid signal (Lock, No
Lock), or if there is a valid and synchronous signal (Sync). The second row shows the sample
frequency measured by the hardware. In Clock Mode the clock reference is shown (Current…).
See also chapter 35.1.
Usually soundcards and audio interfaces generate their internal clock (master mode) by a
quartz. Therefore the internal clock can be set to 44.1 kHz or 48 kHz, but not to a value in between. SteadyClock, RME's sensational Low Jitter Clock System, is based on a Direct Digital
Synthesizer (DDS). This superior circuitry can generate nearly any frequency with highest precision.
DDS has been implemented into the Fireface with regard to the needs of professional video
applications, as well as to maximum flexibility. The section Pitch includes both a list of typical
video frequencies (so called pull up/pull down at 0.1% and 4%) and a fader to freely change the
basic sample rate in steps of 1 Hz (!) over a range of +/- 5%.
The Pitch function requires the Fireface to be in clock mode Master! The frequency setting
will only be applied to this one specific Fireface!
Changing the sample rate during record/playback often results in a loss of audio, or brings
up warning messages of the audio software. Therefore the desired sample rate should be
set at least coarsely before starting the software.
Coarse
Coarse modification in steps of 50
Hz is done by clicking with the
mouse to the left and right of the
fader knob.
Fine
Fine modification in steps of 1 Hz is
done by using the left/right cursor
keys.
Reset
Ctrl key plus left mouse click.
Application examples
Pitch allows for a simultaneous
change of speed and tune during
record and playback. From
alignment to other sources up to
creative effects – everything is
possible.
Pitch enables you to intentionally
de-tune the complete DAW. This
way, the DAW can match
instruments which have a wrong or
unchangeable tuning.
Pitch allows for the change of the sample rate of all WDM devices at the same time. Since Vista
this is no longer possible via the audio program, thus requires a manual reconfiguration of all
WDM devices. Changing the sample rate from the Settings dialog solves this problem. As the
change within the system requires some time, record/playback should not be started immediately, but not before 5 seconds after a change.
Defines the reference level of the rear
analog inputs 5-8.
Line Out
Defines the reference level of the rear
analog outputs 1-6.
Phones
Defines the reference level of the
analog outputs 7/8.
Microphone Inputs
Link 1+2
Ganging of faders 1 and 2. Simplifies
the setup in case both channels shall
have the same setting/values
+48V
Phantom power (48V) can be
selected for each microphone input
separately.
Channel 1 and 2 of the Fireface UC
have digitally controlled microphone
preamps of the highest quality. The
digital control offers a gain setting in steps of 1 dB within a range of 10 dB to 65 dB. The configuration is done either directly at the unit via the rotary encoder, or via the Settings dialog's tab
Gain. The current gain is displayed in dB beside the fader.
In the lower range the fader jumps from 10 dB to 0 dB. This useful additional setting provides
line signal compatibility at up to +10 dBu.
Instrument / Line Inputs
Link 3+4
Ganging of faders 3 and 4. Simplifies the setup in case both channels shall have the same setting/values
Pad
Pad decrease the input sensitivity by 12 dB.
Inst
Activate the option Instrument to use inputs 3 and 4 with instruments. The input impedance is
raised to 470 kOhm, the input sensitivity increases by 10 dB.
The inputs 3 and 4 have a digitally controlled preamp. It allows for an additional gain between 0
and 18 dB, in steps of 0.5 dB. The configuration is done either directly at the unit via the rotary
encoder, or via the Settings dialog's tab Gain. The current gain is displayed in dB beside the
fader. Further information is found in chapter 25.3.
The Fireface UC can extract position information as MTC as well APP (ASIO Positioning Protocol) from Timecode (LTC, SMPTE) received at the analog input 4.
A detected Timecode is shown as time information in the LTC In field. The analog input signal
needs a specific level: slowly reduce the level until the display stumbles or completely fails.
Then increase the level by 10 dB.
The field Input State will show further details of the Timecode.
Basically Timecode can also be used as clock source. However, the calculation of the position
information is less precise then. Recommended is a clocking of the Fireface UC with a clock
signal (for example Word) directly from the device that sends the Timecode.
11.5 Clock Modes - Synchronization
In the digital world, all devices must be either Master (clock source) or Slave (clock receiver)..
Whenever several devices are linked within a system, there must always be a single master
clock. The Fireface UC utilizes a very user-friendly, intelligent clock control, called AutoSync.
In AutoSync mode, the system constantly scans all digital inputs for a valid signal. If any valid
signal is found, the Fireface switches from the internal quartz (Clock Mode – Current Internal) to
a clock extracted from the input signal (Clock Mode – Current ADAT etc). The difference to a
usual slave mode is that whenever the clock reference fails, the system will automatically use
its internal clock and operate in Master mode.
AutoSync guarantees that record and record-while-play will always work correctly. In certain
cases however, e.g. when the inputs and outputs of a DAT machine are connected directly to
the Fireface UC, AutoSync may cause feedback in the digital carrier, so synchronization breaks
down. To remedy this, switch the Fireface’s clock mode to Master (Clock Source – Internal).
A digital system can only have one master! If the Fireface’s clock mode is set to 'Internal',
all other devices must be set to ‘Slave’.
The Fireface UC's ADAT and SPDIF inputs operate simultaneously. Because there is no input
selector, the Fireface UC has to be told which one of the signals is the sync reference (a digital
device can only be clocked from a single source). The Clock Source selection is used to define
a preferred input for the automatic clock system. This input will stay active as long as a valid
signal is found.
To cope with some situations which may arise in studio practice, defining a sync reference is
essential. One example: An ADAT recorder is connected to the ADAT input (ADAT immediately
becomes the AutoSync source) and a CD player is connected to the SPDIF input. Try recording
a few samples from the CD and you will be disappointed - few CD players can be synchronized.
The samples will inevitably be corrupted, because the signal from the CD player is read with the
(wrong) clock from the ADAT i.e. out of sync. In this case, the Clock Source should be set temporarily to SPDIF.
If several digital devices are to be used simultaneously in a system, they not only have to operate with the same sample frequency but also be synchronous with each other. Therefore a Master has to be defined within the digital system, sending the same clock signal to all the other
devices.
RME’s exclusive SyncCheck technology (first implemented in the Hammerfall) enables an easy
to use check and display of the current clock status. Input Status indicates whether there is a
valid signal (Lock, No Lock) for each input (Word Clock, ADAT, SPDIF and LTC), or if there is a
valid and synchronous signal (Sync). In the field Clock Mode the clock reference is shown (Cur-
rent…). See chapter 35.1.
In practice, SyncCheck provides the user with an easy way of checking whether all digital devices connected to the system are properly configured. With SyncCheck, finally anyone can
master this common source of error, previously one of the most complex issues in the digital
studio world.
Note on SPDIF and Word
: Thanks to its lightning fast SteadyClock PLL the Fireface UC is not
only capable of handling standard frequencies, but also any sample rate between 28 and 200
kHz. For varispeed operation the word clock input is the preferred source.
In the audio application being used, Fireface must be selected as output device. This can often
be found in the Options, Preferences or Settings menus as Playback Device, Audio Devices, Audio etc.
We recommend switching all system sounds off (via >Control Panel /Sounds<). Also Fireface
should not be the Preferred Device for playback, as this could cause loss of synchronization
and unwanted noises. If you feel you cannot do without system sounds, you should consider
using the on-board sound device or buying a cheap Blaster clone and select this as Preferred Device in >Control Panel /Multimedia /Audio< or >Control Panel /Sounds /Playback<.
The screenshot shows a typical configuration dialog. After selecting a device, audio data is sent
to an analog or digital (ADAT / SPDIF) port, depending on which has been selected as playback
device.
Increasing the number and/or size of audio buffers may prevent the audio signal from breaking
up, but also increases latency i.e. output is delayed. For synchronized playback of audio and
MIDI (or similar), be sure to activate the checkbox ‘Get position from audio driver’.
Note on Windows Vista
Since Vista the audio application can no longer control the sample rate under WDM. Instead the
user has to work himself through numerous settings (up to 32 with a MADI card!), and to set the
sample rate to the same value per stereo device.
Therefore the driver of the Fireface UC includes a workaround: the sample rate can be set
globally for all WDM devices within the Settings dialog, see chapter 11.1.
When using popular DVD software players like WinDVD and PowerDVD, their audio data
stream can be sent to any AC-3/DTS capable receiver using the Fireface's SPDIF outputs. For
this to work, the WDM SPDIF device of the Fireface UC has to be selected in >Control Panel/ Sounds and Multimedia/ Audio< or >Control Panel/ Sound/Playback<. Also check 'use preferred
device only'.
The DVD software's audio properties now show the options 'SPDIF Out' or similar. When selecting it, the software will transfer the non-decoded digital multichannel data stream to the Fireface.
This 'SPDIF' signal sounds like chopped noise at highest level. Therefore the Fireface automatically sets the non-audio bit within the digital data stream, to prevent most SPDIF receivers
from accepting the signal, and to prevent any attached equipment from being damaged.
Multichannel
PowerDVD and WinDVD can also operate as software decoder, sending a DVD's multichannel
data stream directly to the analog outputs of the Fireface. All modes are supported, from 2 to 8
channels, at 16 bit resolution and up to 192 kHz sample rate. For this to work select
XP: the WDM playback device ’Loudspeaker’ of the Fireface UC in >Control Panel/ Sounds and
Multimedia/ Audio<, and 'Use only default devices' has to be checked. Additionally the loud-speaker setup, found under >Volume/ Speaker Settings/ Advanced< has to be changed from
Stereo to 5.1 Surround.
Vista: the Fireface UC’s WDM playback device ’Loudspeaker’ in >Control Panel/ Sound/ Play-
back < as ‘Standard’. Additionally the loudspeaker setup, found under >Configuration<, has to
be changed from Stereo to 5.1 Surround.
PowerDVD's and WinDVD's audio properties now list several multichannel modes. If one of
these is selected, the software sends the decoded analog multichannel data to the Fireface.
The typical channel assignment for surround playback is:
1 - Left
2 - Right
3 - Center
4 - LFE (Low Frequency Effects)
5 - SL (Surround Left)
6 - SR (Surround Right)
: Selecting the Fireface to be used as system playback device is against our recommen-
Note 1
dations, as professional interfaces should not be disturbed by system events. Make sure to reassign the selection after usage, or to disable any system sounds (tab Sounds, scheme 'No
audio').
Note 2
: The DVD player will be synced backwards from the Fireface. This means when using
AutoSync and/or word clock, the playback speed and pitch follows the incoming clock signal.
The driver offers one WDM streaming device per stereo pair, like ADAT 1+2 (Fireface UC).
WDM Streaming is Microsoft's current driver and audio system, directly embedded into the operating system. WDM Streaming is hardly usable for professional music purposes, as all data is
processed by the so called Kernel Mixer, causing a latency of at least 30 ms. Additionally, WDM
can perform sample rate conversions unnoticed, cause offsets between record and playback
data, block channels unintentionally and much more.
Several programs do not offer any direct device selection. Instead they use the playback device
selected in Windows under
XP: <Control Panel/ Sounds and Multimedia/ Audio>
Vista: <Control Panel/ Sound/ Playback>The program Sonar from Cakewalk is unique in many ways. Sonar uses the so called WDM
Kernel Streaming, bypassing the WDM mixer, thus achieves a similar performance as ASIO
and our MME driver (see below).
Because of the driver's multichannel streaming ability (option Interleaved, see chapter 12.5),
Sonar not only finds the stereo device mentioned above, but also the 8-channel interleaved
devices, and adds the channel number at the end:
Fireface Analog (1+2) is the first stereo device
Fireface Analog (3+4) is the next stereo device
Fireface Analog (1+2) 3/4 are the channels 3/4 of the first 8-channel interleaved device.
It is not recommended to use these special interleaved devices. Also it is not possible to use
one stereo channel twice (the basic and the interleaved device).
Multi-Channel using WDM
The WDM Streaming device Loudspeaker (Analog 1+2) of the RME driver can operate as usual
stereo device, or as up to 8-channel device.
An 8-channel playback using the Windows Media Player requires the speaker setup 7.1 Sur-round. Configure as follows:
The Fireface’s ADAT optical interface offers sample rates of up to 192 kHz using a standard
ADAT recorder. For this to work single-channel data is spread to two or four ADAT channels
using the Sample Multiplexing technique. This reduces the number of available ADAT channels
from 8 to 4 or 2 per ADAT port.
It is nearly impossible to change the number of WDM devices without a reboot of the computer.
Therefore whenever the Fireface UC changes into Double Speed (88.2/96 kHz) or Quad Speed
mode (176.4/192 kHz) all devices stay present, but become partly inactive.
WDM Stereo devices Double Speed Quad Speed
Fireface UC Analog (1+2) Fireface UC Analog (1+2) Fireface UC Analog (1+2)
Fireface UC Analog (3+4) Fireface UC Analog (3+4) Fireface UC Analog (3+4)
Fireface UC Analog (5+6) Fireface UC Analog (5+6) Fireface UC Analog (5+6)
Fireface UC Analog (7+8) Fireface UC Analog (7+8) Fireface UC Analog (7+8)
Fireface UC SPDIF Fireface UC SPDIF Fireface UC SPDIF
Fireface UC AS (1+2) Fireface UC AS (1+2) Fireface UC AS (1+2)
Fireface UC ADAT (3+4) Fireface UC ADAT (3+4) Fireface UC ADAT (3+4)
Fireface UC ADAT (5+6) Fireface UC ADAT (5+6) Fireface UC ADAT (5+6)
Fireface UC ADAT (7+8) Fireface UC ADAT (7+8) Fireface UC ADAT (7+8)
12.5 Multi-client Operation
RME audio interfaces support multi-client operation. This means several programs can be used
at the same time. Also ASIO and WDM can be used simultaneously. The use of multi-client
operation requires to follow two simple rules:
I.e. it is not possible to use one software with 44.1 kHz and the other with 48 kHz.
•Different software can not use the same playback channels at the same time.
If for example Cubase uses channels 1/2, this playback pair can't be used in WaveLab, no matter if ASIO or WDM. However, this is no limitation at all, because TotalMix allows for any output
routing, and therefore a playback of multiple software on the same hardware outputs. Note that
identical inputs can be used at the same time, as the driver simply sends the data to all applications simultaneously.
ASIO-Multiclient
RME audio interfaces support ASIO multi-client operation. It is possible to use more than one
ASIO software at the same time. Again the sample rate has to be identical, and each software
has to use its own playback channels. Again the inputs can be used simultaneously.
RME's sophisticated tool DIGICheck is an exception to this rule. It operates like an ASIO host,
using a special technique to access playback channels already occupied. Therefore DIGICheck
is able to analyse and display playback data from any software, no matter which format the
software uses.
Unlike analog soundcards which produce empty wave files (or noise) when no input signal is
present, digital interfaces always need a valid input signal to start recording.
Taking this into account, RME added two important features to the Fireface UC: a comprehensive I/O signal status display showing sample frequency, lock and sync status in the Settings
dialog, and status LEDs for each input.
The sample frequency shown in the Settings dialog (see chapter 11.1, screenshot Settings) is
useful as a quick display of the current configuration (the board itself and all connected external
equipment). If no sample frequency is recognized, it will read ‘No Lock’.
This way, configuring any suitable audio application for digital recording is simple. After selecting the required input, Fireface UC displays the current sample frequency. This parameter can
then be changed in the application’s audio attributes (or similar) dialog.
It often makes sense to monitor the input signal or send it directly to the output. This can be
done at zero latency using TotalMix (see chapter 29).
An automated control of real-time monitoring can be achieved by Steinberg’s ASIO protocol
with RME’s ASIO 2.2 drivers and all ASIO 2.0 compatible programs. When 'ASIO Direct Monitoring' has been switched on, the input signal is routed in real-time to the output whenever a
recording is started (punch-in).
12.7 Analog Recording
For recordings via the analog inputs the corresponding record device has to be chosen (Fireface UC Analog (x+x)).
The input sensitivity of the rear inputs can be changed via the Settings dialog (Gain/Level) in
three steps so that the recording is done with optimized levels. A further improvement is possible by slowly raising the source’s output level until the peak level meters in TotalMix reach
about –3 dB.
The input sensitivity of the frontside analog inputs can also be adjusted directly at the Fireface
via the rotary encoder knob. A Signal LED and a Clip LED help to find the correct level adjustment.
More information is found in chapter 25.2 and 25.3.
Start the ASIO software and select ASIO Fireface UC as the audio I/O device.
Fireface UC supports
ASIO Direct Monitoring
(ADM). Please note that
currently Nuendo,
Cubase and Logic either
do not support ADM
completely or error-free.
The most often reported
problem is the wrong
behaviour of panorama
in a stereo channel.
When the sample frequency is set to 88.2 or
96 kHz, the number of
ASIO ADAT channels is
reduced to 4. At a sample rate of 176.4 or 192
kHz (Quad Speed
Mode) the ADAT I/O is
no longer available. Nevertheless it will send out a synchronized ADAT signal at a quarter of the
sample rate. The ASIO driver corrects the number of channels when changing from Single to
Double or Quad Speed. The channel situation is explained in chapter 13.2.
Please note that when changing the sample rate range between Single, Double and Quad
Speed the number of channels presented from the ASIO driver will change too. This may require a reset of the I/O list in the audio software.
In case of a drift between audio and MIDI, or in case of a fixed deviation (MIDI notes placed
close before or behind the correct position), the settings in Cubase/Nuendo have to be
changed. At the time of print the option 'Use System Timestamp' should be activated. The Fireface UC supports both MME MIDI and DirectMusic MIDI. It depends on the used application
which one will work better.
At a sample rate of 88.2 or 96 kHz, the ADAT optical input and outputs operate in S/MUX mode,
so the number of available channels per port is reduced from 8 to 4.
At a sample rate of 176.4 and 192 kHz, the ADAT optical input and outputs operate in S/MUX4
mode, so the number of available channels per port is limited to 2.
Please note that when changing the sample rate range between Single, Double and Quad
Speed the number of channels presented from the ASIO driver will change too. This may require a reset of the I/O list in the audio software.
Single Speed Double Speed Quad Speed
Fireface UC Analog 1 to 8 Fireface UC Analog 1 to 8 Fireface UC Analog 1 to 8
Fireface UC SPDIF L / R Fireface UC SPDIF L / R Fireface UC SPDIF L / R
Fireface UC AS 1 to 2 Fireface UC AS 1 to 2 Fireface UC AS 1 to 2
Fireface UC ADAT 3 to 4 Fireface UC ADAT 3 to 4 Fireface UC ADAT 3 to 4
Fireface UC ADAT 5 to 6 Fireface UC ADAT 5 to 6 Fireface UC ADAT 5 to 6
Fireface UC ADAT 7 to 8 Fireface UC ADAT 7 to 8 Fireface UC ADAT 7 to 8
13.3 Known Problems
If a computer does not provide sufficient CPU-power and/or sufficient USB-bus transfer rates,
then drop outs, crackling and noise will appear. Such effects can be avoided by using a higher
buffer setting/latency in the Settings dialog of the Fireface UC. Furthermore PlugIns should be
deactivated temporarily to make sure they do not cause these problems.
More information can be found in chapter 35.3.
Another common source of trouble is incorrect synchronization. ASIO does not support asyn-
chronous operation, which means that the input and output signals not only have to use the
same sample frequency, but also have to be in sync. All devices connected to the Fireface UC
must be properly configured for Full Duplex operation. As long as SyncCheck (in the Settings
dialog) only displays Lock instead of Sync, the devices have not been set up properly!
The same applies when using more than one Fireface UC. They all have to be in sync. Else a
periodically repeated noise will be heard.
The current driver supports up to three Fireface UC. All units have to be in sync, i.e. have to
receive valid sync information (either via word clock or by using AutoSync and feeding synchronized signals).
• If one of the Firefaces is set to clock mode Master, all others have to be set to clock mode
Slave, and have to be synced from the master, for example by feeding word clock. The
clock modes of all units have to be set up correctly in the Fireface Settings dialog.
• If all units are fed with a synchronous clock, i.e. all units show Sync in their Settings dialog,
all channels can be used at once. This is especially easy to handle under ASIO, as the
ASIO driver presents all units as one.
The driver takes care of the numbering of all Firefaces, so that it doesn't change. The unit with
the lowest serial number is always 'Fireface (1)'. Please note:
• If the Fireface (1) is switched off, Fireface (2) logically turns to the first and only Fireface. If
Fireface (1) is switched on later, the numbering changes and the unit becomes Fireface (2)
immediately.
• The driver has no control on the numbering of the WDM devices. Therefore it might happen
that the WDM devices (2) are mapped to unit (1), especially when switching on more Firefaces during a Windows session. A reboot with all Firefaces already operational should
solve this problem.
Note
: TotalMix is part of the hardware of each Fireface. Up to three mixers are available, but
these are separated and can't interchange data. Therefore a global mixer for all units is not
possible.
When using more than one Fireface UC the USB bus might get overloaded. To prevent this
connect all units to different USB busses. This should be possible without further hardware as
many popular USB 2.0 interfaces come as twins, which can be checked in the Device Manager
as follows:
• Connect the Fireface UC to a USB port
• Start the Device Manager, View set to ‘Devices by Connection’
• Expand ACPI x86-based PC, Microsoft ACPI compatible System, PCI bus
Here you usually find two entries of a USB2 Enhanced Host Controller. All USB devices are
connected via a Root Hub, the Fireface UC will show up here too. By a simple reconnection to
other ports this view lets you check to which one of the two controllers the Fireface is connected
to. And with two units, if they use the same controller or not.
This information can also be used to makes sure that an external USB hard drive will not interfere with the Fireface UC, by ensuring its connection to a different bus (controller).
Especially with laptops it can be seen that all internal devices and all sockets are connected to
the same controller, and that the second controller is not used at all. Then all units operate on
the same bus and fight against each other for bandwidth..
The DIGICheck software is a unique utility developed for testing, measuring and analysing digital audio streams. Although this Windows software is fairly self-explanatory, it still includes a
comprehensive online help. DIGICheck 5.2 operates as multi-client ASIO host, therefore can be
used in parallel to any software, with both inputs and outputs (!). The following is a short summary of the currently available functions:
•Level Meter. High precision 24-bit resolution, 2/10/28 channels. Application examples: Peak
level measurement, RMS level measurement, over-detection, phase correlation measurement, dynamic range and signal-to-noise ratios, RMS to peak difference (loudness), long
term peak measurement, input check. Oversampling mode for levels higher than 0 dBFS.
Supports visualization according to the K-System.
•Hardware Level Meter for Input, Playback and Output. Reference Level Meter freely con-
figurable, causing near zero CPU load, because calculated from the Fireface hardware.
•Vector Audio Scope. World wide unique Goniometer showing the typical afterglow of a
oscilloscope-tube. Includes Correlation meter and level meter.
•Surround Audio Scope. Professional Surround Level Meter with extended correlation
analysis.
•Spectral Analyser. World wide unique 10-, 20- or 30-band display in analog bandpass filter
technology. 192 kHz-capable!
•Bit Statistics & Noise. Shows the true resolution of audio signals as well as errors and DC
offset. Includes Signal to Noise measurement in dB and dBA, plus DC measurement.
• Totalyser. Spectral Analyser, Level Meter and Vector Audio Scope in a single window.
• Channel Status Display. Detailed analysis and display of SPDIF and AES/EBU Channel
Status data.
• Global Record. Long-term recording of all channels at lowest system load.
• Completely multi-client. Open as many measurement windows as you like, on any chan-
nels and inputs or outputs!
To install DIGICheck, go to the \DIGICheck directory on the RME Driver CD and run setup.exe.
Follow the instructions prompted on the screen.
DIGICheck is constantly updated. The latest version is always available on our website
www.rme-audio.com, section Downloads/Add-Ons.
The newest information can always be found on our website www.rme-audio.com, section FAQ,
Latest Additions.
Important: Check that the correct (PC = Windows, AP = Mac) firmware is loaded by a doubleclick on the rotary encoder.
The input signal cannot be monitored in real-time
• ASIO Direct Monitoring has not been enabled within the DAW, and/or monitoring has been
disabled globally (TotalMix Options).
The 8 ADAT channels don’t seem to work
• The optical output has been switched to 'SPDIF'. As can be seen in the block diagram, all
channels and their assignments still exist, but the optical transmitter has been disconnected
from ADAT. The ADAT playback devices are still usable by routing and mixing them in TotalMix to other outputs.
Playback works, but record doesn’t
• Check that there is a valid signal at the input. If so, the current sample frequency is displayed in the Settings dialog.
• Check whether the Fireface UC has been selected as recording device in the audio application.
• Check whether the sample frequency set in the audio application (‘Recording properties’ or
similar) matches the input signal.
• Check that cables/devices have not been connected in a closed loop. If so, set the system’s
clock mode to Master.
Crackle during record or playback
• Increase the number and size of buffers in the ‘Settings’ dialog or in the application.
• Try different cables (coaxial or optical) to rule out any defects here.
• Check that cables/devices have not been connected in a closed loop. If so, set the system’s
clock mode to ‘Master’.
• Check the Settings dialog for displayed Errors.
Driver installation and Settings dialog/TotalMix work, but a playback or record is not possible
• While recognition and control of the device are low bandwidth applications, playback/record
needs the full USB transmission performance. Therefore, defective USB cables with limited
transmission bandwidth can cause such errors.
After the flash process the Fireface UC does not start correctly
•Pushing the rotary encoder while switching on the unit will load the Safety BIOS. The flash
process should then be repeated, maybe even on a different computer.
• Connect the power supply to the Fireface and then to any suitable power outlet. Power-on
the Fireface with the rear power switch.
• Check the correct firmware version by a double click on the rotary encoder button. The
display PC means Windows, the display AP means Mac. The double click automatically
switches between both versions. Note: a change of state is only possible when the unit is
not connected to the computer.
• Connect computer and Fireface using the supplied USB cable.
• Mac OS X detects the new hardware as Fireface UC Mac (serial number).
18. Driver
18.1 Driver Installation
After the Fireface has been connected, (see 19. Hardware Installation) install the drivers from
the RME Driver CD. The driver files are located in the folder Fireface UC. Installation works
automatically by a double-click on the file Fireface USB.pkg.
RME recommends downloading the latest driver version from the RME website. If done, the
procedure is as follows:
A double-click onto driver_usb_mac.zip expands the archive file to Fireface USB.pkg. Installation works automatically by a double-click on this file.
During driver installation the programs Fireface USB Settings and Fireface USB Mixer (TotalMix) are copied to the Applications folder. It is recommended to link these two programs to
the Dock so that they are always available.
A reboot of the computer is not required.
Possible reasons why a Fireface UC is not found after driver installation:
• The USB port is not active in the system (check in System Profiler, USB)
• The USB cable is not or not correctly inserted into the socket
• No power. After switching the Fireface on, at least the 7 segment display has to be lit.
In case of a driver update it's not necessary to remove the old driver first, it will be overwritten
during the installation. In case of problems the driver files can be deleted manually by dragging
them to the trash bin:
/Applications/Fireface USB Mixer
/Applications/Fireface USB Settings
/System/Library/Extensions/FirefaceUSB.kext
/Users/username/Library/Preferences/Fireface USB folder
/Users/username/Library/Preferences/de.rme-audio.FirefaceUSBMixer.plist
/Users/username/Library/Preferences/de.rme-audio.Fireface_USB_Settings.plist
18.3 Firmware Update
The Flash Update Tool updates the firmware of the Fireface UC to the latest version. It requires
an already installed driver.
Start the program Fireface USB Flash. The Flash Update Tool displays the current revision of
the Fireface UC's firmware, and whether it needs an update or not. If so, simply press the 'Update' button. A progress bar will indicate when the flash process is finished (Verify Ok).
If more than one Fireface is installed, all units can be flashed by changing to the next tab and
repeating the process.
After the update the unit needs to be reset. This is done by powering down the Fireface for a
few seconds. A reboot of the computer is not necessary.
When the update fails (status: failure), the unit's second BIOS will be used from the next cold
boot on (Secure BIOS Technology). Therefore the unit stays fully functional. The flash process
should then be tried again on a different computer.
In case the automatic activation of the Safety-BIOS fails it can also be activated manually: Push
the rotary encoder button while switching on the unit. This method is also useful to temporarily
deactivate a newer firmware. With pressed button the older version of the firmware will be
loaded (at the time of writing this manual version 82).
Configuring the Fireface UC is done via its own settings dialog. Start the program Fireface USB
Settings. The mixer of the Fireface (TotalMix) can be configured by starting the program Fireface USB Mixer.
The Fireface’s hardware offers a number of helpful, well thought-of practical functions and options which affect how the card operates - it can be configured to suit many different requirements. The following is available in the 'Settings' dialog:
• Level of analog I/Os
• Configuration of digital I/Os
• Synchronization behaviour
• Current sample rate
• State of input and output
Any changes performed in the
Settings dialog are applied
immediately - confirmation (e.g. by
exiting the dialog) is not required.
However, settings should not be
changed during playback or record
if it can be avoided, as this can
cause unwanted noises.
The status displays at the bottom of
the dialog box give precise
information about the current status
of the system, and the status of all
digital signals.
Input Status indicates for each input
(Word, optical, SPDIF coax.)
whether there is a valid signal
(Lock, No Lock), or if there is a valid
and synchronous signal (Sync). The
Clock Mode display shows the clock
Channel 1 and 2 of the Fireface UC have digitally controlled microphone preamps of the highest
quality. The digital control offers a gain setting in steps of 1 dB within a range of 10 dB to 65 dB.
The configuration is done either directly at the unit via the rotary encoder, or via the Settings
dialog's tab Gain. The current gain is displayed in dB beside the fader.
In the lower range the fader jumps from 10 dB to 0 dB. This useful additional setting provides
line signal compatibility at up to +10 dBu.
Phantom power (48V) can be selected for each microphone input separately. Link 1+2 gangs faders 1 and 2. This simplifies the setup in case both channels shall have the
same setting/values.
Instrument / Line Inputs
The inputs 3 and 4 have a digitally controlled preamp. It allows for an additional gain between 0
and 18 dB, in steps of 0.5 dB. The configuration is done either directly at the unit via the rotary
encoder, or via the Settings dialog's tab Gain. The current gain is displayed in dB beside the
fader.
Activate the option Instrument to use inputs 3 and 4 with instruments. The input impedance is
raised to 470 kOhm, the input sensitivity increases by 10 dB. Pad decrease the input sensitivity
by 12 dB.
Link 3+4 gangs faders 3 and 4. This simplifies the setup in case both channels shall have the
same setting/values. More information is found in chapter 25.3.
Level
Line In
Defines the reference level of the rear analog inputs 5-8.
Line Out
Defines the reference level of the rear analog outputs 1-6.
Phones
Defines the reference level of the analog outputs 7/8.
Clock Mode
Sample Rate
Used to set the current sample rate. This is the same setting as in the Audio MIDI Setup, just
added here for your convenience.
At ongoing record/playback the selection is greyed out, so no change is possible.
Clock Source
The unit can be configured to use its own clock (Internal = Master), or one of the input signals
(Word, Optical, SPDIF coax., = Slave). If the selected source isn't available, the unit will change
to the next available one (AutoSync). If none is available then the internal clock is used. The
current clock source is displayed to the right.
Input Status
Indicates for each input (Word, optical, SPDIF coax.) whether there is a valid signal (Lock, No
Lock), or if there is a valid and synchronous signal (Sync). The second row shows the sample
frequency measured by the hardware. In Clock Mode the clock reference is shown (Current…).
See also chapter 35.1.
The word clock output signal usually equals the current sample rate. Selecting Single Speed
causes the output signal to always stay within the range of 32 kHz to 48 kHz. So at 96 kHz and
192 kHz sample rate, the output word clock is 48 kHz.
Optical
The optical TOSLINK output can operate as ADAT or SPDIF output. The Channel Status is
fixed to Consumer state.
: The optical input detects the incoming format automatically.
Note
SPDIF coax.
The coaxial SPDIF output can have the Channel Status Consumer or Professional. For further
details please refer to chapter 27.2.
19.2 Clock Modes - Synchronization
In the digital world, all devices must be either Master (clock source) or Slave (clock receiver)..
Whenever several devices are linked within a system, there must always be a single master
clock. The Fireface UC utilizes a very user-friendly, intelligent clock control, called AutoSync.
In AutoSync mode, the system constantly scans all digital inputs for a valid signal. If any valid
signal is found, the Fireface switches from the internal quartz (Clock Mode – Current Internal) to
a clock extracted from the input signal (Clock Mode – Current ADAT etc). The difference to a
usual slave mode is that whenever the clock reference fails, the system will automatically use
its internal clock and operate in Master mode.
AutoSync guarantees that record and record-while-play will always work correctly. In certain
cases however, e.g. when the inputs and outputs of a DAT machine are connected directly to
the Fireface UC, AutoSync may cause feedback in the digital carrier, so synchronization breaks
down. To remedy this, switch the Fireface’s clock mode over to Master (Clock Source – Internal).
A digital system can only have one master! If the Fireface’s clock mode is set to 'Internal',
all other devices must be set to ‘Slave’.
The Fireface UC's ADAT and SPDIF inputs operate simultaneously. Because there is no input
selector, the Fireface UC has to be told which one of the signals is the sync reference (a digital
device can only be clocked from a single source). The Clock Source selection is used to define
a preferred input for the automatic clock system. This input will stay active as long as a valid
signal is found.
To cope with some situations which may arise in studio practice, defining a sync reference is
essential. One example: An ADAT recorder is connected to the ADAT input (ADAT immediately
becomes the AutoSync source) and a CD player is connected to the SPDIF input. Try recording
a few samples from the CD and you will be disappointed - few CD players can be synchronized.
The samples will inevitably be corrupted, because the signal from the CD player is read with the
(wrong) clock from the ADAT i.e. out of sync. In this case, the Clock Source should be set temporarily to SPDIF.
If several digital devices are to be used simultaneously in a system, they not only have to operate with the same sample frequency but also be synchronous with each other. Therefore a Master has to be defined within the digital system, sending the same clock signal to all the other
devices.
RME’s exclusive SyncCheck technology (first implemented in the Hammerfall) enables an easy
to use check and display of the current clock status. Input Status indicates whether there is a
valid signal (Lock, No Lock) for each input (Word Clock, ADAT, SPDIF and LTC), or if there is a
valid and synchronous signal (Sync). In the field Clock Mode the clock reference is shown (Cur-
rent…). See chapter 35.1.
In practice, SyncCheck provides
the user with an easy way of
checking whether all digital devices
connected to the system are
properly configured. With
SyncCheck, finally anyone can
master this common source of
error, previously one of the most
complex issues in the digital studio
world.
Note on SPDIF and Word
: Thanks
to its lightning fast SteadyClock
PLL the Fireface UC is not only
capable of handling standard
frequencies, but also any sample
rate between 28 and 200 kHz. For
varispeed operation the word clock
input is the preferred source.
The driver with the file suffix zip provided by RME is a compressed archive. Zip is directly supported by OS X, a double click on the file is all one needs to do.
The driver consists of a package file (pkg). A double click will start the OS X installer.
The actual audio driver appears as a kernel extension file. The installer copies it to >System/
Library/ Extensions<. Its name is FirefaceUSB.kext. It is visible in the Finder, allowing you to
verify date and driver version. Yet, in fact this again is a folder containing subdirectories and
files.
Nonetheless, this 'driver file' can be removed by simply dragging it to the trash bin. This can be
helpful in case a driver installation fails.
20.2 MIDI doesn't work
In some cases the applications do not show the MIDI port. The reason for this is usually visible
within the Audio MIDI Setup. It displays no RME MIDI device, or the device is greyed out and
therefore inactive. Mostly, removing the greyed out device and searching for MIDI devices again
will solve the problem.
The Fireface UC is class compliant. Therefore it comes without a driver. OS X recognizes it as
MIDI device and will be using it with the driver included in the operating system.
20.3 Repairing Disk Permissions
Repairing permission can solve problems with the installation process - plus many others. To do
this, launch Disk Utility located in Utilities. Select your system drive in the drive/volume list to
the left. The First Aid tab to the right now allows you to check and repair disk permissions.
20.4 Supported Sample Rates
RME's Mac OS X driver supports all sampling frequencies provided by the hardware. This includes 32 kHz and 64 kHz, and even 128 kHz, 176.4 kHz and 192 kHz for the analog and
SPDIF I/Os.
But not any software will support all the hardware's sample rates. The hardware's capabilities
can easily be verified in the Audio MIDI Setup. Select Audio devices under Properties of:
and choose the Fireface UC. A click on Format will list the supported sample frequencies.
The Fireface’s ADAT optical interface offers sample rates of up to 192 kHz using a standard
ADAT recorder. For this to work single-channel data is spread to two or four ADAT channels
using the Sample Multiplexing technique. This reduces the number of available ADAT channels
from 8 to 4 or 2 per ADAT port.
It is not possible to change the number of Core Audio devices without a reboot of the computer.
Therefore whenever the Fireface UC changes into Double Speed (88.2/96 kHz) or Quad Speed
mode (176.4/192 kHz) all devices stay present, but become partly inactive.
Single Speed Double Speed Quad Speed
Fireface UC Analog 1 to 8 Fireface UC Analog 1 to 8 Fireface UC Analog 1 to 8
Fireface UC SPDIF L / R Fireface UC SPDIF L / R Fireface UC SPDIF L / R
Fireface UC ADAT 1 to 2 Fireface UC ADAT 1 to 2 Fireface UC ADAT 1 to 2
Fireface UC ADAT 3 to 4 Fireface UC ADAT 3 to 4 Fireface UC ADAT 3 to 4
Fireface UC ADAT 5 to 6 Fireface UC ADAT 5 to 6 Fireface UC ADAT 5 to 6
Fireface UC ADAT 7 to 8 Fireface UC ADAT 7 to 8 Fireface UC ADAT 7 to 8
20.6 Various Information
The driver of the Fireface UC requires at least Mac OS 10.5, as special USB functions are used
that are not available in older versions of the operating system.
Via >System Preferences/ Audio-MIDI Setup< the hardware can be configured for the system
wide usage. Programs that don't support card or channel selection will use the device selected
as Standard-Input and Standard-Output. (Soundstudio, Mplayer, Amplitube etc.).
In the lower part of the window, the audio hardware's capabilities are shown and can be
changed in some cases. On the record side no changes are possible. Programs that don't support channel selection will always use channels 1/2, the first stereo pair. To access other inputs,
use the following workaround with TotalMix: route the desired input signal to output channels
1/2. Hold the Ctrl key down and click on the labels AN1 and AN2 in the third row. Their labels
turn red, the internal loop mode is active. Result: the desired input signal is now available at
input channel 1/2, without further delay/latency.
Use Speaker Setup to freely configure the playback to all available channels. Even multichan-
nel playback (Surround, DVD Player) can be set up this way.
Multicard Operation
OS X supports more than one audio device. Since 10.4 (Tiger) Core Audio offers the function
Aggregate Devices, which allows to combine several devices into one, so that a multi-device
operation is possible with any software.
The Fireface driver adds a number to each unit, so they are fully accessible in any multicardcapable software.
OS X erlaubt die Verwendung von mehr als einem Audiogerät und deren gleichzeitige Nutzung
in einem Programm. Dies geschieht über die Funktion Aggregate Devices, mit dem sich mehrere Geräte zu einem zusammenfassen lassen.
The current driver supports up to three Fireface UC. All units have to be in sync, i.e. have to
receive valid sync information (either via word clock or by using AutoSync and feeding synchronized signals).
• If one of the Firefaces is set to clock mode Master, all others have to be set to clock mode
Slave, and have to be synced from the master, for example by feeding word clock. The
clock modes of all units have to be set up correctly in the Fireface Settings dialog.
• If all units are fed with a synchronous clock, i.e. all units show Sync in their Settings dialog,
all channels can be used at once.
When using more than one Fireface UC the USB bus might get overloaded. To prevent this
connect all units to different USB busses.
Note
: TotalMix is part of the hardware of each Fireface. Up to three mixers are available, but
these are separated and can't interchange data. Therefore a global mixer for all units is not
possible.
22. DIGICheck Mac
The DIGICheck software is a unique utility developed for testing, measuring and analysing digital audio streams. Although this Windows software is fairly self-explanatory, it still includes a
comprehensive online help. DIGICheck 0.5 operates in parallel to any software, showing all
input data. The following is a short summary of the currently available functions:
level measurement, RMS level measurement, over-detection, phase correlation measurement, dynamic range and signal-to-noise ratios, RMS to peak difference (loudness), long
term peak measurement, input check. Oversampling mode for levels higher than 0 dBFS.
Supports visualization according to the K-System.
•Hardware Level Meter for Input, Playback and Output. Reference Level Meter freely con-
figurable, causing near zero CPU load, because calculated from the Fireface hardware.
•Vector Audio Scope. World wide unique Goniometer showing the typical afterglow of a
oscilloscope-tube. Includes Correlation meter and level meter.
•Surround Audio Scope. Professional Surround Level Meter with extended correlation
analysis.
•Spectral Analyser. World wide unique 10-, 20- or 30-band display in analog bandpass filter
technology. 192 kHz-capable!
• Totalyser. Spectral Analyser, Level Meter and Vector Audio Scope in a single window.
• Completely multi-client. Open as many measurement windows as you like, on any chan-
nels and inputs or outputs!
To install DIGICheck, go to the \DIGICheck directory on the RME Driver CD and run setup.exe.
Follow the instructions prompted on the screen.
DIGICheck is constantly updated. The latest version is always available on our website
www.rme-audio.com, section Downloads/Add-Ons.
The newest information can always be found on our website www.rme-audio.com, section FAQ,
latest Additions.
Important: Check that the correct (PC = Windows, AP = Mac) firmware is loaded by a doubleclick on the rotary encoder.
The unit and drivers have been installed correctly, but playback does not work:
• Is Fireface UC listed in the System Profiler? (Vendor ID 2613).
• Has Fireface been selected as current playback device in the audio application?
The 8 ADAT channels don’t seem to work
• The optical output has been switched to 'SPDIF'. As can be seen in the block diagram, all
channels and their assignments still exist, but the optical transmitter has been disconnected
from ADAT. The ADAT playback devices are still usable by routing and mixing them in TotalMix to other outputs.
Playback works, but record doesn’t:
• Check that there is a valid signal at the input. If so, the current sample frequency is displayed in the Settings dialog.
• Check whether the Fireface UC has been selected as recording device in the audio application.
• Check whether the sample frequency set in the audio application (‘Recording properties’ or
similar) matches the input signal.
• Check that cables/devices have not been connected in a closed loop. If so, set the system’s
clock mode to ‘Master’.
Crackle during record or playback:
• Increase the number and size of buffers in the application.
• Try different cables (coaxial or optical) to rule out any defects here.
• Check that cables/devices have not been connected in a closed loop. If so, set the system’s
clock mode to ‘Master’.
Possible causes for a Fireface not working
• The USB cable is not, or not correctly inserted into the socket
• No power. After switching the Fireface on, at least the red Host error LED has to be lit.
Driver installation and Settings dialog/TotalMix work, but a playback or record is not possible
• While recognition and control of the device are low bandwidth applications, playback/record
needs the full FireWire transmission performance. Therefore, defective FireWire cables with
limited transmission bandwidth can cause such an error scheme.
After the flash process the Fireface UC does not start correctly
•Pushing the rotary encoder while switching on the unit will load the Safety BIOS. The flash
process should then be repeated, maybe even on a different computer.
The Fireface UC has an internal memory to permanently store all configuration data. The data
is stored directly after any change, and are loaded at power-on. Saved settings are:
Settings dialog
Sample rate, clock mode Master/Slave, configuration of the channels and the digital I/Os.
TotalMix
The complete mixer state.
This also improves the clock situation immediately after power-on, avoiding wrong clocking and
noise disturbances in a complex setup, caused by wrong synchronization. Usually the unit will
be configured by the Windows or Mac driver, so for the time between power-on of the computer
up to the loading of the driver its state might be wrong.
This total configuration feature in stand-alone operation - without any connected computer turns the Fireface into lots of dedicated devices, see examples in chapter 24. Furthermore TotalMix (and with this all application examples) can be MIDI controlled even in stand-alone operation, see chapter 32.7, Stand-Alone MIDI Control.
24.1 Front Panel Operation
The knob on the front, a so called rotary encoder, serves to set the input gains and output volumes directly at the unit. The encoder operates either in CHANNEL or in LEVEL mode. Pushing
the knob changes between these modes. The currently active mode is indicated by a green
LED.
In CHANNEL mode, selection of the desired channel is done by turning
the knob. The following strings will be shown in the display:
i.1 bis i.4 Mic input 1 up to instrument/line input 4
L.1 bis L.6 Line output 1 up to line output 6
PH Phones (line output 7/8)
SP SPDIF output
A.1 bis A.8 ADAT output 1 up to 8
i.1 / i.2
The gain of the two microphone inputs 1/2 can be defined in the range of 10 dB up to 65 dB in
steps of 1 dB. Additionally the setting 0 dB is available. The gain change is performed in analog
domain in hardware.
i.3 / i.4
The gain of the two instrument/line inputs 3/4 can be defined in the range of 0 dB up to 18 dB in
steps of 0.5 dB. The x.5 dB values are signalled by a dot to the right. The gain change is performed in analog domain in hardware.
L.1 bis L.6, PH, SP, A.1 bis A.8
The output levels of these outputs can be defined in the range +6 dB down to –58 dB in steps of
1 dB. Additionally the setting maximum attenuation (Mute) is available. The gain change is performed digitally by TotalMix.
Stereo Mode
Pushing the knob for more than a second activates the Link (Gang) mode. The display will show
off or on. In stereo (on) mode, the display only presents the left channels of a stereo pair (L1,
L3, L5...). The gain and volume setting is then valid for both channels.
When loading TotalMix' factory default 1 into the unit, the Fireface becomes a high quality 8channel AD/DA-converter, which also provides a monitoring of all 8 DA-channels via channels
7/8 (Preset 2: also monitoring all 8 inputs). A small modification allows for a monitoring of all
I/Os via the SPDIF I/O.
24.3 2-Channel Mic Preamp
Use TotalMix to route the two microphone inputs directly to the analog outputs. This turns the
Fireface UC into a 2-channel microphone preamp. The AD- and DA-conversion will cause a
small delay of the signals of around 0.35 ms (at 192 kHz, see chapter 35.2). But this is not
really relevant, as it is the same delay that would be caused by changing the microphone's position by about 12 centimeter (5 inches).
24.4 Monitor Mixer
TotalMix allows ANY configuration of all I/Os of the Fireface. For example, set up the device as
monitor mixer for 8 analog signals, 8 digital via ADAT and 2 via SPDIF. Additionally, TotalMix
lets you set up ANY submixes, so all existing outputs can be used for different and independent
monitorings of the input signals. The perfect headphone monitor mixer!
24.5 Digital Format Converter
As TotalMix allows for any routing of the input signals, the Fireface UC can be used as ADAT to
SPDIF converter, and SPDIF to ADAT converter.
24.6 Analog/digital Routing Matrix
The Matrix in TotalMix enables you to route and link all inputs and outputs completely freely. All
the above functionalities are even available simultaneously, can be mixed and combined in
many ways. Simply said: the Fireface UC is a perfect analog/digital routing matrix!
The Fireface has balanced line inputs as 1/4" TRS jacks on the back of the unit. The electronic
input stage is built in a servo balanced design which handles unbalanced (mono jacks) and
balanced (stereo jacks) correctly, automatically adjusting the level reference.
When using unbalanced cables with TRS jacks: be sure to connect the 'ring' contact of the
TRS jack to ground. Otherwise noise may occur, caused by the unconnected negative input
of the balanced input.
One of the main issues when working with an AD-converter is to maintain the full dynamic
range within the best operating level. Therefore the Fireface UC internally uses hi-quality electronic switches, which allow for a perfect adaptation of all rear inputs to the three most often
used studio levels.
The 'standardized' studio levels do not result in a (often desired) full scale level, but take some
additional digital headroom into consideration. The amount of headroom is different in different
standards, and again differently implemented by different manufacturers. Because of this we
decided to define the levels of the Fireface in the most compatible way.
Reference 0 dBFS @ Headroom
Lo Gain +19 dBu 15 dB
+4 dBu +13 dBu 9 dB
-10 dBV +2 dBV 12 dB
With +4 dBu selected, the according headroom meets the latest EBU recommendations for
Broadcast usage. At -10 dBV a headroom of 12 dB is common practice, each mixing desk operating at -10 dBV is able to send and receive much higher levels. Lo Gain is best suited for
professional users who prefer to work balanced and at highest levels. Lo Gain provides 15 dB
headroom at +4 dBu nominal level.
The above levels are also found in our ADI-8 series of AD/DA converters, the Multiface, and
even in our Mic-Preamps QuadMic and OctaMic. Therefore all RME devices are fully compatible to each other.
25.2 Microphone / Line Front
The balanced microphone inputs of the Fireface UC offer an adjustable gain of 10 dB up to 65
dB. The soft switching, hi-current Phantom power (48 Volt) provides a professional handling of
condensor mics. The usage of a hi-end integrated circuit (PGA 2500) guarantees outstanding
sound quality, stunning low THD, and maximum Signal to Noise ratio in any gain setting.
The two combo jacks also allow for the usage of mono and stereo TRS jacks. These jacks usually carry line level signals. To adress this correctly, the input gain can be lowered to 0 dB. The
inputs are still operating servo-balanced, but can now handle levels up to +10 dBu.
Overall the inputs 1/2 can work with levels from –55 dBu up to +10 dBu. Two LEDs display a
present signal (from –65 dBFS on) and warn against overload (-2 dBFS).
The instrument inputs 3/4 of the Fireface UC are exceptionally flexible. Several gain and impedance options make them the ideal receivers for both line and instrument signals.
Line
Inputs 3/4 have balanced line inputs as 1/4" TRS jacks. The electronic input stage is built in a
servo balanced design which handles unbalanced (mono jacks) and balanced (stereo jacks)
correctly, automatically adjusting the level reference.
When using unbalanced cables with TRS jacks: be sure to connect the 'ring' contact of the
TRS jack to ground. Otherwise noise may occur, caused by the unconnected negative input
of the balanced input.
Inputs 3/4 operate in perfect harmony with the rear inputs 5 to 8, as they use the same level
references:
Reference 5-8 0 dBFS @ Setting 3/4 Input Gain 3/4
Lo Gain +19 dBu Pad +6 dB
+4 dBu +13 dBu No Pad 0 dB
-10 dBV +2 dBV Inst 0 dB
- +8 dBu Inst + Pad 0 dB
Instrument
The main difference between a line- and instrument input is the input's impedance. Via the option Inst 3 and Inst 4 in the Settings dialog, the input impedance changes from 10 kOhm to 470
kOhm. At the same time the input sensitivity rises by 10 dB. Using active instruments this setting can already cause overload. Therefore the option Pad is still available, reducing the sensitivity by 12 dB.
The instrument input operates servo-balanced, thus handles balanced signals correctly as well.
Input Gain
Via Input Gain in the Settings dialog, or using the rotary encoder on the front, an additional gain
can be applied to the input signals of channels 3/4. The gain range is 0 dB up to 18 dB, in steps
of 0.5 dB. This option not only allows for the use of sources with low level output signals, but for
example also to adjust the balance between channel 3 and 4 precisely, already before the recording takes place.
Overall the inputs 3/4 can work with levels from –16 dBu up to +25 dBu. Two LEDs display a
present signal (from –65 dBFS on) and warn against overload (-2 dBFS).
The eight short circuit protected, low impedance line outputs are available as 1/4" TRS jacks on
the back of the unit. The electronic output stage is built in a servo balanced design which handles unbalanced (mono jacks) and balanced (stereo jacks) correctly.
To maintain an optimum level for devices connected to the analog outputs, the Fireface UC
internally uses hi-quality electronic switches, which allow for a perfect adaptation of all outputs
to the three most often used studio levels.
As with the analog inputs, the analog output levels are defined to maintain a problem-free operation with most other devices. The headroom of the Fireface UC lies between 9 and 15 dB,
according to the chosen reference level:
Reference 0 dBFS @ Headroom
Hi Gain +19 dBu 15 dB
+4 dBu +13 dBu 9 dB
-10 dBV +2 dBV 12 dB
With +4 dBu selected, the according headroom meets the latest EBU recommendations for
Broadcast usage. At -10 dBV a headroom of 12 dB is common practice, each mixing desk operating at -10 dBV is able to send and receive much higher levels. Lo Gain is best suited for
professional users who prefer to work balanced and at highest levels. Lo Gain provides 15 dB
headroom at +4 dBu nominal level.
The above levels are also found in our ADI-8 series of AD/DA converters, the Multiface, and
even in our Mic-Preamps QuadMic and OctaMic. Therefore all RME devices are fully compatible to each other.
26.2 Phones (7/8)
Channels 7/8 of the Fireface are available on the front via one 1/4" unbalanced TRS jack (stereo output). These channels use the same converters as the other line outputs, therefore offer
the same technical specifications.
These outputs are special low impedance types, ready to be used with headphones. But they
can also be used as high-quality (yet unbalanced) line outputs. To optimize this stereo output
for monitoring purposes, outputs 7/8 have their own 3-stage hardware-based reference level
circuitry, providing perfect pre-adjustment for the connected monitor speakers. The main volume control is then performed via TotalMix, either at the computer or directly at the unit (rotary
encoder, PH).
In case the output should operate as
line output, an adapter TRS plug to
RCA phono plugs, or TRS plug to TS
plugs is required.
The pin assignment follows international standards. The left channel is
connected to the tip, the right channel
to the ring of the TRS jack/plug.
The ADAT optical input of the Fireface UC is fully compatible with all ADAT optical outputs.
RME's unsurpassed Bitclock PLL prevents clicks and drop outs even in extreme varipitch operation, and guarantees a fast and low jitter lock to the digital input signal. A usual TOSLINK
cable is sufficient for connection. More information on Double Speed (S/MUX) can be found in
chapter 35.4.
ADAT In
Interface for a device sending an ADAT signal to the Fireface UC. Carries the channels 1 to 8.
When receiving a Double Speed signal, this input carries the channels 1 to 4, at Quad Speed
the inputs 1 and 2.
ADAT Out
Interface for a device receiving an ADAT signal from the Fireface UC. Transmits channels 1 to
8. When sending a Double Speed signal, this port carries channels 1 to 4, at Quad Speed the
channels 1 and 2.
27.2 SPDIF
The Fireface UC has two SPDIF inputs and outputs. Coaxial and optical can be used simultaneously, with different audio signals. Note that the source have to be synchronous, as the unit
does not have a dedicated sample rate converter.
The optical input automatically switches to SPDIF operation when such a signal is detected.
The audio information is then shown in TotalMix on the first two ADAT channels, In 11 and In
12.
To send out SPDIF from the optical output, select the option Optical – SPDIF in the Settings
dialog. The output signal in TotalMix has to be present on the first ADAT channels, AS 1+2.
The Fireface UC’s inputs accepts SPDIF as well as AES/EBU.
To receive signals in AES/EBU format,
an adapter cable is required. Pins 2 and
3 of a female XLR plug are connected
individually to the two pins of a phono
plug. The cable shielding is connected
to pin 1 of the XLR and the ground
contact of the phono plug.
Special Characteristics of the SPDIF Output
Apart from the audio data itself, digital signals in SPDIF or AES/EBU format have a header containing channel status information. False channel status is a common cause of malfunction. The
Fireface UC ignores the received header and creates a totally new one for the output signal.
The Fireface UC’s new output header is optimized for largest compatibility with other digital
devices:
• 32 kHz, 44.1 kHz, 48 kHz, 88.2 kHz, 96 kHz, 176.4 kHz, 192 kHz depending on the current
sample rate
• Audio use, Non-Audio
• No Copyright, Copy Permitted
• Format Consumer or Professional
• Category General, Generation not indicated
• 2-channel, No Emphasis or 50/15 µs
• Aux bits Audio Use
Professional AES/EBU equipment can be connected to the Fireface UC thanks to its ‘Professional’ format option and a doubled output voltage. Output cables should have the same pinout
as those used for input (see above), but with a male XLR plug instead of a female one.
Note that most consumer HiFi equipment (with optical or phono SPDIF inputs) will only
accept signals in ‘Consumer’ format!
Therefore in SPDIF mode the optical output is set fixed to Consumer.
27.3 MIDI
Fireface UC offers two MIDI I/O via four 5-pin DIN jacks. The MIDI ports are added to the system by the driver. Using MIDI capable software, these ports can be accessed under the name
Fireface UC Midi. Using more than one Fireface, the operating system adds a consecutive
number to the port name, like Fireface UC MIDI (2) etc.
The MIDI ports support multi-client operation. A MIDI input signal can be received from several
programs at the same time. Even the MIDI output can be used by multiple programs simultaneously. However, due to the limited bandwidth of MIDI, this kind of application will often show
various problems.
Note
: The MIDI input LED displays any kind of MIDI activity, including MIDI Clock, MTC and
Active Sensing. The latter is sent by most keyboards every 0.3 seconds.
SteadyClock guarantees an excellent performance in all clock modes. Based on the highly efficient jitter suppression, the Fireface refreshes and cleans up any clock signal, and provides it
as reference clock at the BNC output (see section 35.8).
Input
The Fireface's word clock input is active when Word has been selected as Clock Source in the
Settings dialog and a valid word clock signal is present. The signal at the BNC input can be
Single, Double or Quad Speed, the Fireface UC automatically adapts to it. As soon as a valid
signal is detected, the WC LED is lit, and the Settings dialog shows either Lock or Sync (see
chapter 35.1).
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 Fireface's 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 on the back beside the word clock
input socket. Use a small pencil or similar and carefully push the blue switch so that it snaps into its lock
position. The yellow LED will be lit when termination is
active. Another push will release it again and deactivate the termination.
Output
The word clock output of the Fireface is constantly active, providing the current sample frequency as word clock signal. As a result, in Master mode the provided word clock is defined by
the currently used software. In Slave mode the provided frequency is identical to the one present at the currently chosen clock input. When the current clock signal fails, the Fireface UC
switches to Master mode and adjusts itself to the next, best matching frequency (44.1 kHz, 48
kHz etc.).
Selecting Single Speed in the Settings dialog causes the output signal to always stay within the
range of 32 kHz to 48 kHz. So at 96 kHz and 192 kHz sample rate, the output word clock is 48
kHz.
The received word clock signal can be distributed to other devices by using the word clock output. With this the usual T-adapter can be avoided, and the Fireface UC operates as Signal Re-fresher. This kind of operation is highly recommended, because
• input and output are phase-locked and in phase (0°) to each other
• SteadyClock removes nearly all jitter from the input signal
• the exceptional input (1 Vpp sensitivity instead of the usual 2.5 Vpp, dc cut, Signal Adapta-
tion Circuit) plus SteadyClock guarantee a secure function even with highly critical word
clock signals
Thanks to a low impedance, but short circuit proof output, the Fireface delivers 4 Vpp to 75
Ohms. For wrong termination with 2 x 75 Ohms (37.5 Ohms), there are still 3.3 Vpp at the output.
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 and ADAT 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!
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 typically 15 times 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 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 Fireface UC.
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. You will find all the necessary components (T-adapters, terminators, cables) in most electronics and/or computer stores.
The latter usually carries 50 Ohms components. The 75 Ohms 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 bigger
studios.
The Fireface's word clock input can be high-impedance or terminated internally, ensuring
maximum flexibility. If termination is necessary (e.g. because the Fireface is the last device in
the chain), push the switch at the back beside the BNC socket (see chapter 28.1).
In case the Fireface UC 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.
Due to the outstanding SteadyClock technology of the Fireface UC, we recommend not to
pass the input signal via T-adapter, but to use the Fireface's word clock output instead.
Thanks to SteadyClock, the input signal will both be freed from jitter and - in case of loss or
drop out – be reset to a valid frequency.
28.4 Operation
The green Lock LED on the front (STATE) will light up as soon as a word clock signal is detected. To change to word clock as clock source, activate clock mode AutoSync and switch
Pref. Sync Ref to Word Clock within the Settings dialog. The status display AutoSync Ref
changes to Word as soon as a valid signal is present at the BNC jack. This message has the
same meaning as the green Lock LED, but appears on the monitor, i.e. the user can check
immediately whether a valid word clock signal is present and is currently being used.
The Input State also displays the frequency of the current word clock signal, measured by the
hardware.
The Fireface UC includes a powerful digital real-time mixer, the Fireface mixer, based on RME’s
unique, sample-rate independent TotalMix technology. It allows for practically unlimited mixing
and routing operations, with all inputs and playback channels simultaneously, to any hardware
outputs.
Here are some typical applications for TotalMix:
• Setting up delay-free submixes (headphone mixes). The Fireface allows for up to 9 (!) fully
independent stereo submixes. On an analog mixing desk, this would equal 18 (!) Aux sends.
• Unlimited routing of inputs and outputs (free utilisation, patchbay functionality).
• Distributing signals to several outputs at a time. TotalMix offers state-of-the-art splitter and
distributor functions.
• Simultaneous playback of different programs using only one stereo output. The ASIO multi-
client driver allows to use several programs at the same time, but only on different playback
channels. TotalMix provides the means to mix and monitor these on a single stereo output.
• Mixing of the input signal to the playback signal (complete ASIO Direct Monitoring). RME not
only is the pioneer of ADM, but also offers the most complete implementation of the ADM
functions.
• Integration of external devices. Use TotalMix to insert external effects devices, be it in the
playback or in the record path. Depending on the current application, the functionality equals
insert or effects send and effects return, for example as used during real-time monitoring
when adding some reverb to the vocals.
Every single input channel, playback channel and hardware output features a Peak and RMS
level meter, calculated in hardware. These level displays are very useful to determine the presence and routing destinations of the audio signals.
For a better understanding of the TotalMix mixer you should know the following:
• As shown in the block diagram (next page), t he record signal usually stays un-altered. To-
talMix does not reside within the record path, and does not change the record level or the
audio data to be recorded (exception: loopback mode).
• The hardware input signal can be passed on as often as desired, even with different levels.
This is a big difference to conventional mixing desks, where the channel fader always controls the level for all routing destinations simultaneously.
• The level meter of inputs and playback channels are connected pre-fader, to be able to
visually monitor where a signal is currently present. The level meters of the hardware’s outputs are connected post-fader, thus displaying the actual output level.
The visual design of the TotalMix mixer is a result of its capability to route hardware inputs and
software playback channels to any hardware output. The Fireface UC provides 18 input channels, 18 software playback channels, and 18 hardware output channels:
36 channels don't fit on the screen side by side, neither does such an arrangement provide a
useful overview. The input channel should be placed above the corresponding output channel.
Therefore, the channels have been arranged as known from an Inline desk, so that the row
Software Playback equals the Tape Return of a real mixing desk:
• Top row: Hardware inputs. The level shown is that of the input signal, i. e. fader independ-
ent. Via fader and routing field, any input channel can be routed and mixed to any hardware
output (bottom row).
• Middle row: Playback channels (playback tracks of the audio software). Via fader and routing
field, any playback channel can be routed and mixed to any hardware output (third row).
•Bottom row (third row): Hardware outputs. Here, the total level of the output can be adjusted.
This may be the level of connected loudspeakers, or the necessity to reduce the level of an
overloaded submix.
The following chapters explain step by step all functions of the user interface.
A single channel consists of various elements:
Input channels and playback channels each have a mute and solo button.
Below there is the pan pot, realized as indicator bar (L/R) in order to save space.
In the field below, the present level is displayed in RMS or Peak, being updated
about every half a second. Overs (overload) are indicated here by an additional red
dot.
Next is the fader with a level meter. The meter shows both peak values (zero attack,
1 sample is enough for a full scale display) by means of a yellow line, and
mathematically correct RMS values by means of a green bar. The RMS display has a
relatively slow time constant, so that it shows the average loudness quite well.
Below the fader, the current gain and panorama values are shown.
The grey area shows the channel name. Selecting one or more channels is done by clicking on
the grey label which turns orange then. A click in the third row with pressed Ctrl-key activates
internal loopback mode, the label turns red. A right mouse click opens a dialog to type in a new
name.
The black area (routing field) shows the current routing target. A mouse click opens the routing
window to select a routing target. The list shows all currently activated routings by checkmarks
in front of the routing targets.
29.4 Tour de TotalMix
This chapter is a practical guide and introduction on how to use TotalMix and on how TotalMix
works.
Starting up TotalMix the last settings are recalled automatically. When executing the application
for the first time, a default file is loaded, sending all playback tracks 1:1 to the corresponding
hardware outputs with 0 dB gain, and activating phones monitoring.
Hold down Ctrl and click on preset button 1 to make sure that factory preset 1 is loaded. The
faders in the top row are set to maximum attenuation (called m.a. in the following), so there is
no monitoring of the input channels. The Submix View is active, therefore for improved overview all outputs except Phones are greyed out. Additionally all faders are set to the routing target Phones. All faders of the middle row are set to 0 dB, so no matter on which channels a
playback happens, the audio will be audible via the Phones output. Just try it!
We will now create a submix on analog outputs 1/2. Please start a multitrack playback. In the
third row, click on the channels of hardware output AN1 or AN2. The Submix View changes
from Phones to AN1/AN2. Both the fader settings and the output levels of all other channels are
still visible, but greyed out for improved orientation.
As soon as AN1/AN2 became active, all faders of the second row jumped to their bottom position – except those of playback channel 1/2. This is correct, because as mentioned above the
factory preset includes a 1:1 routing. Click on AN3/AN4 and the faders above are the only active ones, same for AN5/AN6 and so on.
Back to AN1/AN2. Now you can change all the faders of all inputs and playback channels just
as you like, thus making any input and playback signals audible via the outputs AN1/AN2. The
panorama can be changed too. Click into the area above the fader and drag the green bar in
order to set the panorama between left and right. The level meters of the third row display the
level changes in real-time.
Overall it is very easy to set up a submix for any output: select
output channel, set fader and pans of inputs and playbacks – ready!
For advanced users sometimes it makes sense to work without
Submix View. Example: you want to see and set up some channels
of different submixes simultaneously, without the need to change
between them all the time. Switch off the Submix View by a click on
the green button. Now the black routing fields below the faders no
longer show the same entry (AN1+2), but completely different ones.
The fader and pan position is the one of the individually shown routing destination.
In playback channel 1 (middle row), labelled Out 1, click onto the
routing field below the label. A list pops up, showing a checkmark in
front of 'AN 1+2' and 'Phones'. So currently playback channel 1 is
sent to these two routing destinations. Click onto 'AN 5+6'. The list
disappears, the routing field no longer shows 'AN1+2', but 'AN 5+6'.
Now move the fader with the mouse. As soon as the fader value is
unequal m.a., the present state is being stored and routing is activated. Move the fader button to around 0 dB. The present gain value
is displayed below the fader in green letters.
In the lower row, on channel 5, you can see the level of what you
are hearing from output 5. The level meter of the hardware output
shows the outgoing level. Click into the area above the fader and
drag the mouse in order to set the panorama, in this case the routing
between channels 5 and 6. The present pan value is also being
displayed below the fader.
Please carry out the same steps for Out 2 now, in order to route it to
output 6 as well.
In short: While editing the Submix AN5/AN6 you have direct access to other submixes on other
channels, because their routing fields are set to different destinations. And you get a direct view
of how their faders and panoramas are set up.
This kind of visual presentation is a mighty one, but for many users it is hard to understand, and it requires a deep understanding of complex routing visualizations. Therefore
we usually recommend to work with Submix View.
Often signals are stereo, i. e. a pair of two channels. It is therefore helpful to be able to make
the routing settings for two channels at once. Hold down the Ctrl-key and click into the routing
field of Out 3. The routing list pops up with a checkmark at 'AN 3+4'. Select 'AN 5+6'. Now, Out
4 has already been set to 'AN 5+6' as well.
When you want to set the fader to exactly 0 dB, this can be difficult, depending on the mouse
configuration. Move the fader close to the 0 position and now press the Shift-key. This activates
the fine mode, which stretches the mouse movements by a factor of 8. In this mode, a gain
setting accurate to 0.1 dB is no problem at all.
Please set Out 4 to a gain of around -20 dB and the pan close to center. Now click onto the
routing field. You'll now see two checkmarks, one at 'AN 3+4', the other one at 'AN 5+6'. Click
onto 'SPDIF'. The window disappears, fader and pan jump to their initial values, the signal can
now be routed to the SPDIF output. You can continue like this until all entries have got a
checkmark, i. e. you can send the signal to all outputs simultaneously.
You will certainly have noticed that the signal at the outputs 5/6 did not change while you were
routing channel 4 to other outputs and setting different gain values for those. With all analog
and most digital mixing desks, the fader setting would affect the level for every routed bus - not
so for TotalMix. TotalMix allows for setting all fader values individually. Therefore the faders and
the panpots jump to the appropriate setting as soon as another routing is chosen.
Sometimes you will want the routings not to be independent. Let's say you have sent a signal to
several submixes, and now want to change the signal's volume a bit on all these submixes.
Dragging the faders by use of the right mouse button activates Post Send mode and causes all
routings of the current input or playback channel to be changed in a relative way. Please note
that the fader settings of all routings are memorized. So when pulling the fader to the bottom
(maximum attenuation), the individual settings are back when you right click the mouse and pull
the fader up. The individual settings get lost in m.a. position as soon as the fader is clicked with
the left mouse button. As long as no single level is at m.a. position, the left mouse button can be
used to change the current routing's gain.
The checkmarks are un-checked by moving the fader to m.a. This setting deactivates the routing...why route if there is no level? Click onto 'AN 5+6' in the routing window, pull the fader
down, open the routing window again - the checkmark is gone.
The number of ADAT channels is reduced automatically when entering Double Speed mode (96
kHz). The display is adjusted accordingly, and all fader settings remain stored. Changing into
Quad Speed mode (192 kHz) six ADAT channels vanish. TotalMix then displays a total of only
12 channels.
29.5 Submix View
Such a wide range of possibilities make it difficult to maintain the overview. Because as shown
practically all hardware outputs can be used for different submixes, (up to 9 completely independent stereo submixes, 4 4-channel submixes etc.). And when opening the routing windows
you might see an army of checkmarks, but you don't get an overview, i.e. how the signals come
together and where. This problem is solved by Submix View mode. In this mode, all routing
fields jump to the routing pair just being selected. You can then see immediately, which channels, which fader and pan settings make a submix (for example 'AN 5+6'). At the same time the
Submix View simplifies setting up the mixer, as all channels can be set simultaneously to the
same routing destination with just one click.
Changing to a different destination (output channel) is done in any routing field, or by a click on
the desired output pair in the bottom row.
29.6 Mute and Solo
Mute operates pre-fader, thus mutes all currently active routings of the channel. As soon as any
Mute button is pressed, the Mute Master button lights up in the Quick Access area. With this all
selected mutes can be switched off and on again. You can comfortably set up a mute-group or
activate and deactivate several Mutes simultaneously.
The same holds true for the Solo and the Solo Master buttons. As with conventional mixing
desks, Solo operates only for the output defined as Monitor Main, as a solo-in-place, post
fader. As soon as one Solo button is pressed, the Solo Master button lights up in the Quick
Access area. With this all selected Solos are switched off and on again. You can comfortably
set up a solo-group or activate and deactivate several Solos simultaneously.
This section includes additional options, further improving the handling of TotalMix. The Master
buttons for Mute and Solo have already been described, they allow for group-based working
with these functions.
In the View section the single mixer rows can be made visible or invisible. If the inputs are not
needed for a pristine playback mix, the whole upper vanishes after a click on the Input button. If
the hardware outputs are of no interest either, the window can thus be reduced to the playback
channels to save space. All combinations are possible and allowed.
As described earlier, Submix sets all routing windows to the same selection. Deactivating
Submix automatically recalls the previous view. The mixer can be made smaller horizontally
and vertically. This way TotalMix can be made substantially smaller and space-saving on the
desktop/screen, if you have to monitor or set up only a few channels or level meters.
The Presets are one of the mightiest and most useful features of TotalMix.
Behind the eight buttons, eight files are hidden (see next chapter). These
contain the complete mixer state. All faders and other settings follow the
changing of preset(s) in real-time, just by a single mouse click. The Save
button allows for storing the present settings in any preset. You can change
back and forth between a signal distribution, complete input monitoring, a
stereo and mono mix, and various submixes without any problem.
If any parameter is being altered after loading a preset (e. g. moving a fader),
the preset display flashes in order to announce that something has been
changed, still showing which state the present mix is based on.
If no preset button is lit, another preset had been loaded via the File menu,
Open file. Mixer settings can of course be saved the usual way, and can also
have long file names.
Instead of single presets a complete bank of (8) presets can be loaded at once.
Advantage: The names defined for the preset buttons will be stored and loaded
automatically.
Up to three Firefaces can be used simultaneously. The Unit buttons switch between the devices. Holding down Ctrl while clicking on button Unit 2 or Unit 3 will open another TotalMix
window.
29.8 Presets
TotalMix includes eight factory presets, stored within the program. The user presets can be
changed at any time, because TotalMix stores and reads the changed presets from the files
preset11.umx to preset81.umx, located in Windows' hidden directory Documents and Set-
tings, <Username>, Local Settings, Application Data, RME TotalMix. On the Mac the location is
in the folder User, <Username>, Library / Preferences / Fireface. The first number indicates the
current preset, the second number the current unit.
This method offers two major advantages:
• Presets modified by the user will not be overwritten when reinstalling or updating the driver
• The factory presets remain unchanged, and can be reloaded any time.
Mouse: The original factory presets can be reloaded by holding down the Ctrl-key and clicking
on any preset button. Alternatively the files described above can be renamed, moved to a different directory, or being deleted.
Keyboard: Using Ctrl and any number between 1 and 8 (not on the numeric
keypad!) will load the corresponding factory default preset. The key Alt will
load the user presets instead.
When loading a preset file, for example 'Main Monitor AN 1_2 plus
headphone mix 3_4.umx', the file name will be displayed in the title bar of the
TotalMix window. Also when loading a preset by the preset buttons the name
of the preset is displayed in the title bar. This way it is always clear what the
current TotalMix state is based on.
The eight factory presets offer a pretty good base to modify them to your personal needs. In all
factory presets Submix View is active by default.
Preset 1
Description: All channels routed 1:1, monitoring of all playback channels via Phones.
Details: All inputs maximum attenuation. All playback channels 0 dB, routed to the same output.
All outputs 0 dB, Phones –6 dB. Submix of all inputs and playbacks to channel 7/8 (Phones).
Level display RMS +3 dB. View Submix active.
: This preset is Default, offering the standard functionality of a I/O-system.
Note
Preset 2
Description: All channels routed 1:1, input and playback monitoring via Phones. As Preset 1,
plus submix of all inputs (0 dB) to channels 7/8 (Phones).
Preset 3
Description: All channels routed 1:1, input and playback monitoring via Phones and outputs. As
Preset 2, but all inputs set to 0 dB (1:1 monitoring).
Preset 4
Description: All channels routed 1:1, input and playback monitoring via Phones and outputs. As
Preset 3, but all inputs muted.
Preset 5
Description: Playback monitoring to Phones. As Preset 1, but all outputs except channels 9/10
(Phones) set to maximum attenuation.
Preset 6
Description: All channels routed 1:1, monitoring of all playback channels via Phones and
SPDIF. As Preset 1, plus submix of all playbacks to SPDIF.
Preset 7
Description: Monitoring of all playback channels via Phones and of all input and playback channels via SPDIF. As Preset 2, plus submix of all inputs to SPDIF.
Preset 8
Description: Panic. As Preset 4, but playback channels muted too (no output signal).
Preset Banks
Instead of a single preset, all eight presets can be stored and loaded at once. This is done via
Menu File, Save All Presets as and Open All Presets (file suffix .fpr). After the loading the
presets can be activated by the preset buttons. In case the presets have been renamed (see
chapter 29.11), these names will be stored and loaded too.
The Monitor panel provides several options usually found on analog mixing desks. It offers
quick access to monitoring functions which are needed all the time in typical studio work.
Monitor Main
Use the drop down menu to select the hardware outputs where your main monitors are connected to.
Dim
A click on this button will lower the volume of the Monitor Main output by an
amount set up in the Preferences dialog (see below). This is the same as
moving the third row faders down a bit, but much more convenient, as the old
setting is back by a simple mouse click.
Mono
Sets the stereo output defined above to monaural playback. Useful to check for
mono compatibility and phase problems.
Talkback
A click on this button will dim all signals on the Monitor Phones outputs by an
amount set up in the Preferences dialog. At the same time the control room's
microphone signal (source defined in Preferences) is sent to the three
destinations Monitor Phones. The microphone level is adjusted with the chan-
nel's input fader.
Monitor Phones 1/2/3
Use the drop down menu to select the hardware outputs where the submixes are sent to. These
submixes are usually phones mixdowns for the musicians. A click on the button allows for the
monitoring of the specific submix via the Monitor Main output. So when setting up or modifying
the submix for the musician this process can be monitored easily and any time from the operator.
29.10 Preferences
The dialog box Preferences is available via the
menu Options or directly via F3.
Talkback
Input: Select the input channel of the Talkback
signal (microphone in control room).
Dim: Amount of attenuation of the signals routed to
the Monitor Phones in dB.
Listenback
Input: Select the input channel of the Listenback
signal (microphone in recording room).
Dim: Amount of attenuation of the signals routed to
the Monitor Main in dB.
: The Mute button of the Talkback and
Note
Listenback channel is still active. Therefore it is not
necessary to select <NONE>, in case one of both
shall be deactivated.
Dim: Amount of attenuation of the Monitor Main output in dB. Activated by the Dim button in the
Monitor panel.
Stereo Pan Law
The Pan Law can be set to -6 dB, -4.5 dB, -3 dB and 0 dB. The value chosen defines the level
attenuation in pan center position. This setting is useful because the ASIO host often supports
different pan laws too. Selecting the same value here and in the ASIO host, ASIO Direct Monitoring works perfectly, as both ASIO host and TotalMix use the same pan law. Of course, when
not using ADM it can be changed to a setting different from the factory preset of –6 dB as well.
You will most probably find that -3 dB gives a much more stable loudness when moving an object between left and right.
29.11 Editing the Names
The channel names shown in the grey label area can be
edited. A right mouse click on the grey name field brings up
the dialog box Enter Name. Any name can be entered in
this dialog. Enter/Return closes the dialog box, the grey
label now shows the first letters of the new name. ESC
cancels the process and closes the dialog box.
Moving the mouse above the label
brings up a tool tip with the complete
name.
The hardware outputs (third row) can be edited in the
same way. In this case, the names in the routing drop
down menus will change automatically. Additionally
the names in the drop down menus of the Monitor
section will change as well.
The preset buttons can get
meaningful names in the same
way. Move the mouse above a
preset button, a right mouse click
will bring up the dialog box.
Note that the name shows up as tool tip only as soon as the mouse stays
above the preset button.
The preset button names are not stored in the preset files, but globally in the registry, so won't
change when loading any file or saving any state as preset. Loading a preset bank will update
the names too (see chapter 29.8).
In many situations TotalMix can be controlled quickly and comfortably by the keyboard, making
the mixer setup considerably easier and faster. The Shift-key for the fine mode for faders and
pan pots has already been mentioned. The Ctrl-key can do far more than changing the routing
pairwise:
• Clicking anywhere into the fader area with the Ctrl-key pressed, sets the fader to 0 dB.
• Clicking anywhere into the pan area with the Ctrl-key pressed, sets the panorama to <C>
meaning Center.
• Clicking a preset button while holding down Ctrl, the original factory preset will be loaded.
• Using Ctrl and any number between 1 and 8 (not on the numeric keypad!) will load the cor-
responding factory default preset. Alt plus number loads the user preset.
• Using multiple Firefaces, clicking the button Unit 2 while holding down Ctrl opens a second
TotalMix window for the second Fireface UC, instead of replacing the window contents.
The faders can also be moved pairwise, corresponding to the stereo-routing settings. This is
achieved by pressing the Alt-key and is especially comfortable when setting the SPDIF and
Phones output level. Even the panoramas can be operated with Alt, from stereo through mono
to inversed channels, and also the Mute and Solo buttons (ganged or inversed switching!).
At the same time, TotalMix also supports combinations of these keys. If you press Ctrl and Alt
at the same time, clicking with the mouse makes the faders jump to 0 dB pairwise, and they can
be set pairwise by Shift-Alt in fine mode.
Also very useful: the faders have two mouse areas. The first area is the fader button, which can
be grabbed at any place without changing the current position. This avoids unwanted changes
when clicking onto it. The second area is the whole fader setting area. Clicking into this area
makes the fader jump to the mouse at once. If for instance you want to set several faders to
m.a., it is sufficient to click onto the lower end of the fader path. Which happens pairwise with
the Alt-key pressed.
Using the hotkeys I, O and P the complete row of Input, Playback and Output channels each
can be toggled between visible and invisible. Hotkey S switches Submix view on/off. Those four
hotkeys have the same functionality as the buttons in the View section of the Quick Access
Panel. The Level Meter Setup dialog can be opened via F2 (as in DIGICheck). The dialog box
Preferences is opened via F3.
Hotkey M toggles Mute Master on/off (and with this performs a global mute on/off). Hotkey X
toggles the Matrix view on/off (see chapter 30), hotkey T the mixer view. Hotkey L links all faders as stereo pairs.
Further hotkeys are available to control the configuration of the Level Meters (see chapter
29.14):
Key 4 or 6: Display range 40 or 60 dB
Key E or R: Numerical display showing Peak or RMS
Key 0 or 3: RMS display absolute or relative to 0 dBFS
Always on Top: When active (checked) the TotalMix window will always be on top of the Win-
dows desktop.
: This function may result in problems with windows containing help text, as the TotalMix
Note
window will even be on top of those windows, so the help text isn't readable.
Deactivate Screensaver: When active (checked) any activated Windows screensaver will be
disabled temporarily.
Ignore Position: When active, the windows size and position stored in a file or preset will not
be used. The routing will be activated, but the window will not change.
Ignore I/O Labels: When active the channel names saved in a preset or file will not be loaded,
instead the current ones will be retained.
ASIO Direct Monitoring (Windows only): When de-activated any ADM commands will be
ignored by TotalMix. In other words, ASIO Direct Monitoring is globally de-activated.
Link Faders: Selecting this option all faders will be treated as stereo pairs and moved pairwise. Hotkey L.
MS Processing: Macro for a quick configuration of routing and phase for Mid/Side encoding
and decoding. See chapter 31.7.
Level Meter Setup: Configuration of the Level Meters. Hotkey F2. See chapter 29.14.
Level Meter Text Color: Color adjustment for the Gain and Level meter text displays.
Preferences: Opens a dialog box to configure several functions, like Pan Law, Dim, Talkback
Dim, Listenback Dim. See chapter 29.10.
Enable MIDI Control: Turns MIDI control on. The channels which are currently under MIDI
control are indicated by a colour change of the info field below the faders, black turns to yellow.
Deactivate MIDI in Background: Disables the MIDI control as soon as another application is in
the focus, or in case TotalMix has been minimized.
The Fireface UC calculates all the display values Peak, Over and RMS in hardware, in order to
be capable of using them independent of the software in use, and to cause no CPU load.
Tip: This feature, the Hardware Level Meter, is used by DIGICheck (see chapter 15/22) to
display Peak/RMS level meters of all channels without CPU load.
The level meters integrated in TotalMix - considering their size - cannot be compared with
DIGICheck. Nevertheless they already include many useful functions.
Peak and RMS is displayed for every channel. 'Level Meter Setup' (menu Options or F2) and
direct keyboard hotkeys offer various options:
• Display range 40 or 60 dB (hotkey 4 or 6)
• Release time of the Peak display (Fast/Medium/Slow)
• Numerical display selectable either Peak or RMS (Hotkey E or R)
• Number of consecutive samples for Overload display (1 to 15)
• RMS display absolute or relative to 0 dBFS (Hotkey 3 or 0)
The latter is a point often overlooked, but nonetheless important. A RMS measurement shows
3 dB less for sine signals. While this is
mathematically correct, it is not very reasonable
for a level meter. Therefore the RMS readout is
usually corrected by 3 dB, so that a full scale
sine signal shows 0 dBFS on both Peak and
RMS meters. This setting also yields directly
readable signal-to-noise values. Otherwise the
value shown with noise is 3 dB better than it
actually is (because the reference is not 0 dB,
but -3 dB).
The value displayed in the text field is independent of the setting 60/40 dB, it represents
the full 24 bit range of the RMS measurement.
An example: An RME ADI-8 QS connected to
the Fireface's ADAT port will yield a display of
around -114 dBFS on all eight channel's input
level meters.
This level display of TotalMix also provides means for a constant monitoring of the signal quality. Thus it can be a valuable tool for sound optimization and error removal in the studio.
Measuring SNR (Signal to Noise) is best done with RME’s free software DIGICheck. The
function Bit Statistic includes three different RMS meters for exactly this purpose (RMS
unweighted, A-weighted and DC).
The mixer window of TotalMix looks and operates similar to mixing desks, as it is based on a
conventional stereo design. The matrix display presents a different method of assigning and
routing channels, based on a single channel or monaural design. The matrix view of the Fireface UC has the look and works like a conventional patchbay, adding functionality way beyond
comparable hardware and software solutions. While most patchbays will allow you to connect
inputs to outputs with just the original level (1:1, or 0 dB, as known from mechanical patchbays),
TotalMix allows you to use a freely definable gain value per crosspoint.
Matrix and TotalMix are different ways of displaying the same processes. Because of this both
views are always fully synchronized. Each change in one view is immediately reflected in the
other view as well.
30.2 Elements of the Matrix View
The visual design of the TotalMix Matrix is mainly determined by the architecture of the Fireface
UC:
• Horizontal labels: All hardware outputs
• Vertical labels: All hardware inputs. Below are all
playback channels (software playback channels)
• Green 0.0 dB field: Standard 1:1 routing
• Black gain field: Shows the current gain value as dB
• Orange gain field: This routing is muted
• Blue field: Phase 180° (inverted)
To maintain overview when the window size has been reduced, the left and upper labels are
floating. They won't leave the visible area when scrolling.
30.3 Operation
Using the Matrix is a breeze. It is very easy to indentify the current crosspoint, because the
outer labels light up in orange according to the mouse position.
If input 1 is to be routed to output 1, use the mouse and click one time on crosspoint In 1 / AN 1.
The green 0.0 dB field pops in, another click removes it. To change the gain (equals the use of
a different fader position, see simultaneous display of the mixer view), hold Ctrl down and drag
the mouse up or down, starting from the gain field. The value within the field changes accordingly. The corresponding fader in the mixer view is moving simultaneously, in case the currently
modified routing is visible.
Note the difference between the left side, representing the inputs and software playback channels, and the upper side, representing the hardware outputs.
A gain field marked orange indicates activated mute status. Mute can only be changed in the
mixer view.
A blue field indicates phase inversion. This state is displayed in the Matrix only, and can only be
changed within the Matrix view. Hold down the Shift-key while clicking on an already activated
field. Mute overwrites the phase display, blue becomes orange. If mute is deactivated the phase
inversion is indicated again.
30.4 Advantages of the Matrix
The Matrix not always replaces the mixer view, but it significantly enhances the routing capabilities and - more important - is a brilliant way to get a fast overview of all active routings. It shows
you in a glance what's going on. And since the Matrix operates monaural, it is very easy to set
up specific routings with specific gains.
Example 1: You want TotalMix to route all software outputs to all corresponding hardware outputs, and have a submix of all inputs and software outputs on the Phones output (equals factory
preset 2). Setting up such a submix is easy. But how to check at a later time, that all settings
are still exactly the way you wanted them to be, not sending audio to a different output?
The most effective method to check a routing in mixer view is the Submix View, stepping
through all existing software outputs, and having a very concentrated look at the faders and
displayed levels of each routing. That doesn't sound comfortably nor error-free, right? Here is
where the Matrix shines. In the Matrix view, you simply see a line from upper left to lower right,
all fields marked as unity gain. Plus two rows vertically all at the same level setting. You just
need 2 seconds to be sure no unwanted routing is active anywhere, and that all levels match
precisely!
Example 2: The Matrix allows you to set up routings which would be nearly impossible to
achieve by fiddling around with level and pan. Let's say you want to send input 1 to output 1 at 0
dB, to output 2 at -3 dB, to output 3 at -6 dB and to output 4 at -9 dB. Each time you set up the
right channel (2/4), the change in pan destroys the gain setting of the left channel (1/2). A real
hassle! In Matrix view, you simply click on the corresponding routing point, set the level via Ctrlmouse, and move on. You can see in TotalMix view how pan changes to achieve this special
gain and routing when performing the second (fourth...) setting.
31. TotalMix Super-Features
31.1 ASIO Direct Monitoring (Windows only)
Start Samplitude, Sequoia, Cubase or Nuendo and TotalMix. Activate ADM (ASIO Direct Monitoring), and move a fader in the ASIO host. Now watch the corresponding fader in TotalMix
magically move too. TotalMix reflects all ADM gain and pan changes in realtime. Please note
that faders only move when the currently activated routing (currently visible routing) corresponds to the one in the ASIO host. Also note that the Matrix will show any change, as it shows
all possible routings in one view.
With this TotalMix has become a wonderful debugging tool for ADM. Just move the host's fader
and pan, and see what kind of ADM commands TotalMix receives.
The hardware output row faders are included in all gain calculations, in every possible way.
Example: you have lowered the output level of a submix, or just a specific channel, by some dB.
The audio signal passed through via ADM will be attenuated by the value set in the third row.
Click on the white name label of channel 1 and 2 in TotalMix. Be sure to have channel 3's fader
set to a different position and click on its label too. All three labels have changed to the colour
orange, which means they are selected. Now moving any of these faders will make the other
faders move too. This is called 'building a group of faders', or ganging faders, maintaining their
relative position.
Building groups or ganging can be done in any row, but is limited to operate horizontally within
one row. If you usually don't need this, you can at least gang the analog outputs. The advantage over holding the Alt-key is that Alt sets both channels to the same level (can be handy too),
while grouping via selection will retain any offset (if you need one channel to be louder all the
time etc.).
Note
: The relative positions are memorized until the faders are pulled down so that they reach
upper or lower maximum position and the group is changed (select another channel or deselect
one of the group).
31.3 Copy Routings to other Channels
TotalMix allows to copy complete routing schemes of inputs and outputs.
Example 1: You have input 1 (guitar) routed within several submixes/hardware outputs (=
headphones). Now you'll get another input with keyboards that should appear in the same way
on all headphones. Select input 1, open the menu Edit. It shows 'Copy In 1'. Now select the
desired new input, for example In 8. The menu now shows 'Paste In 1 to In 8'. Click on it - done.
If you are familiar with this functionality just use Ctrl-C and Ctrl-V. Else the self updating menu
will always let you know what actually will happen.
Tip: Have the Matrix window open as second window when doing this. It will show the new routings immediately, so copying is easier to understand and to follow.
Example 2: You have built a comprehensive submix on outputs 4/5, but now need the exact
same signal also on the outputs 6/7. Click on Out 4, Ctrl-C, click on Out 6, Ctrl-V, same with 5/7
- you're done!
The Matrix shows you the difference between both examples. Example 1 means copying lines
(horizontally), while example 2 means copying rows (vertically).
Example 3: Let's say the guitarist finished his recording, and you now need the same signal
again on all headphones, but this time it comes from the recording software (playback row). No
problem, you can even copy between rows 1 and 2 (copying between row 3 and 1/2 isn't possible).
But how to select while a group is active? De-selecting the group first? Not necessary! TotalMix
always updates the copy and paste process with the last selection. This way you don't have to
de-activate any group-selections when desiring to perform a copy and paste action.
31.4 Delete Routings
The fastest way to delete complex routings: select a channel in the mixer view, click on the
menu entry Edit and select Delete. Or simply hit the Del-key. Attention: there is no undo in To-
talMix, so be careful with this function!
TotalMix supports a routing of the subgroup outputs (=hardware outputs, bottom row) to the
recording software. Instead of the signal at the hardware input, the signal at the hardware output is sent to the record software. This way, complete submixes can be recorded without an
external loopback cable. Also the playback from a software can be recorded by another software.
To activate this function, click on the white label in the third row while holding down the Ctrl-key.
The label's colour changes to red. In case the channel has already been part of a group, the
colour will change from yellow to orange, signalling that the group functionality is still active for
this channel.
In loopback mode, the signal at the hardware input of the corresponding channel is no longer
sent to the recording software, but still passed through to TotalMix. Therefore TotalMix can be
used to route this input signal to any hardware output. Using the subgroup recording, the input
can still be recorded on a different channel.
As each of the 18 hardware outputs can be routed to the record software, and none of these
hardware inputs get lost, TotalMix offers an overall flexibility and performance not rivalled by
any other solution.
Additionally the risk of feedbacks, a basic problem of loopback methods, is highly reduced, because the feedback can not happen within the mixer, but only when the audio software is
switched into monitoring mode. The block diagram shows how the software's input signal is
played back, and fed back from the hardware output to the software input. A software monitoring on the subgroup record channels is allowed only as long as the monitoring is routed in both
software and TotalMix to a different channel than the active subgroup recording one.
In real world application, recording a software's output with another software will show the following problem: The record software tries to open the same playback channel as the playback
software (already active), or the playback one has already opened the input channel which
should be used by the record software.
This problem can easily be solved. First make sure that all rules for proper multi-client operation
are met (not using the same record/playback channels in both programs). Then route the playback signal via TotalMix to a hardware output in the range of the record software, and activate it
via Ctrl-mouse for recording.
Mixing several input signals into one record channel
In some cases it is useful to record several sources into only one track. For example when using two microphones recording instruments and loudspeakers. TotalMix' Loopback mode saves
an external mixing desk. Simply route/mix the input signals to the same output (third row), then
re-define this output into a record channel via Ctrl-mouse – that's it. This way any number of
input channels from different sources can be recorded into one single track.
31.6 Using external Effects Devices
With TotalMix a usage of external hardware - like effects devices - is easy and flexible.
Example 1: The singer (microphone input channel 1) shall have some reverb on his head-
phones (outputs 7/8). A direct routing In 1 to Out 7/8 for monitoring had been set up already.
The external reverb is connected to a free output, for example channel 4. In active mode Submix View click on channel 4 in the bottom row. Drag the fader of input 1 to about 0 dB and the
panorama fully to the right. Adjust the input level at the reverb unit to an optimal setting. Next
the output of the reverb unit is connected to a free stereo input, for example 5/6. Use the TotalMix level meters to adjust a matching output level at the reverb unit. Now click on channels
7/8 in the bottom row, and move the fader of inputs 5/6 until the reverb effect gets a bit too loud
in the headphones. Now click on channel 4 in the bottom row again and drag input fader 1 down
a bit until the mix of original signal and reverb is perfect for the singer.
The described procedure is completely identical to the one when using an analog mixing desk.
There the signal of the singer is sent to an output (usually labelled Aux), from there to a reverb
unit, sent back from the reverb unit as stereo wet signal (no original sound), back in through a
stereo input (e.g. Effect return) and mixed to the monitoring signal. The only difference: The Aux
sends on mixing desks are post-fader. Changing the level of the original signal causes a
change of the effects level (here the reverb) too, so that both always have the same ratio.
Tip: Such functionality is available in TotalMix via the right mouse button! Dragging the faders
by use of the right mouse button causes all routings of the current input or playback channel to
be changed in a relative way. This completely equals the function Aux post fader.
Example 2: Inserting an effects device can be done as above, even within the record path.
Other than in the example above the reverb unit also sends the original signal, and there is no
routing of input 1 directly to outputs 7/8. To insert an effects device like a Compressor/Limiter
directly into the record path, the input signal of channel 1 is sent by TotalMix to any output, to
the Compressor, back from the Compressor to any input. This input is now selected within the
record software.
Unfortunately, very often it is not possible within the record software to assign a different input
channel to an existing track 'on the fly'. The loopback mode solves this problem elegantly. The
routing scheme stays the same, with the input channel 1 sent to any output via TotalMix, to the
Compressor, from the Compressor back to any input. Now this input signal is routed directly to
output 1, and output 1 is then switched into loopback mode via Ctrl-mouse.
As explained in chapter 31.5, the hardware input of channel 1 now no longer feeds the record
software, but is still connected to TotalMix (and thus to the Compressor). The record software
receives the signal of submix channel 1 instead – the Compressor's return path.
31.7 MS Processing
The mid/side principle is a special positioning technique for microphones, which results in a mid
signal on one channel and a side signal on the other channel. These information can be transformed back into a stereo signal quite easily. The process sends the monaural mid channel to
left and right, the side channel too, but phase inverted (180°) to the right channel. For a better
understanding: the mid channel represents the function L+R, while the side channel represents
L-R.
During record the monitoring needs to be
done in 'conventional' stereo. As TotalMix
can invert the phase, it also offers the
functionality of a M/S-decoder. The menu
Options includes a macro to simplify the
setup. First select the two input channels,
in the picture to the right Analog In 3 and
4, having the current routing destination
Analog Out 1+2. Now the string MS
Processing In 3+4 to AN 1+2 On is
shown in Options.
After a mouse click TotalMix sets gains and pans correctly. Of course
these settings can also be performed manually. Repeat the last step
to remove all routings (menu Options ...Off).
The M/S-Processing automatically operates as M/S encoder or decoder, depending on the
source signal format. When processing a usual stereo signal, all monaural information will be
shifted into the left channel, all stereo information into the right channel. Thus the stereo signal
is M/S encoded. This yields some interesting insights into the mono/stereo contents of modern
music productions. Additionally some very interesting methods of manipulating the stereo base
and generating stereo effects come up, as it is then very easy to process the side channel with
Low Cut, Expander, Compressor or Delay. The most basic application is already available directly in TotalMix: Changing the level of the side channel allows to manipulate the stereo width
from mono to stereo up to extended, stepless and in real-time.
TotalMix can be remote controlled via MIDI. It is compatible to the widely spread Mackie Control
protocol, so TotalMix can be controlled with all hardware controllers supporting this standard.
Examples are the Mackie Control, Tascam US-2400 or Behringer BCF 2000.
Additionally, the stereo output faders (lowest row) which are set up as MonitorMain outputs in
the Monitor panel can also be controlled by the standard Control Change Volume via MIDI channel 1. With this, the main volume of the Fireface is controllable from nearly any MIDI
equipped hardware device.
32.2 Mapping
TotalMix supports the following Mackie Control surface elements*:
Element: Meaning in TotalMix:
Channel faders 1 – 8 volume
Master fader Main Monitor channel's faders
SEL(1-8) + DYNAMICS reset fader to Unity Gain
V-Pots 1 – 8 pan
pressing V-Pot knobs pan = center
CHANNEL LEFT or REWIND move one channel left
CHANNEL RIGHT or FAST FORWARD move one channel right
BANK LEFT or ARROW LEFT move eight channels left
BANK RIGHT or ARROW RIGHT move eight channels right
ARROW UP or Assignable1/PAGE+ move one row up
ARROW DOWN or Assignable2/PAGE- move one row down
EQ Master Mute
PLUGINS/INSERT Master Solo
STOP Dim Main Monitor
PLAY Talkback
PAN Mono Main Monitor
MUTE Ch. 1 – 8 Mute
SOLO Ch. 1 – 8 Solo
SELECT Ch. 1 – 8 Select
REC Ch. 1 – 8 in Submix mode only: select output bus
F1 - F8 load preset 1 - 8
F9 select Main Monitor
F10 - F12 Monitor Phones 1 - 3
*Tested with Behringer BCF2000 Firmware v1.07 in Mackie Control emulation for Steinberg mode and with Mackie
Control under Mac OS X.
• Open the Preferences dialog (menu Options or F3). Select the MIDI Input and MIDI Output
port where your controller is connected to.
• When no feedback is needed (when using only standard MIDI commands instead of Mackie
Control protocol) select NONE as MIDI Output.
• Check Enable MIDI Control in the Options menu.
32.4 Operation
The channels being under MIDI control are indicated by a colour change of the info field below
the faders, black turns to yellow.
The 8-fader block can be moved horizontally and vertically, in steps of one or eight channels.
Faders can be selected to gang them.
In Submix View mode, the current routing destination (output bus) can be selected via REC Ch.
1 – 8. This equals the selection of a different output channel in the lowest row by a mouse click
when in Submix View. In MIDI operation it is not necessary to jump to the lowest row to perform
this selection. This way even the routing can be easily changed via MIDI.
Full LC Display Support: This option in Preferences (F3) activates complete Mackie Control
LCD support with eight channel names and eight volume/pan values.
Attention: this feature puts a heavy load on the MIDI port when ganging more than 2 faders! In such a case, or when using the Behringer BCF2000, turn off this option.
When Full LC Display Support is turned off, only a brief information about the first fader of the
block (channel and row) is sent. This brief information is also available on the LED display of
the Behringer BCF2000.
Tip for Mac OS X users: LC Xview (www.opuslocus.com
emulating the hardware displays of a Logic/Mackie Control, for use with controllers that can
emulate a Logic/Mackie Control but do not have a display. Examples include the Behringer
BCF2000 and Edirol PCR-series.
Deactivate MIDI in Background (menu Options) disables the MIDI control as soon as another
application is in the focus, or in case TotalMix has been minimized. This way the hardware controller will control the main DAW application only, except when TotalMix is in the foreground.
Often the DAW application can be set to become inactive in background too, so that MIDI control is switched between TotalMix and the application automatically when switching between
both applications.
TotalMix also supports the 9th fader of the Mackie Control. This fader (labelled Master) will control the stereo output faders (lowest row) which are set up as Monitor Main outputs in the Moni-
tor panel. Always and only.
The stereo output faders (lowest row) which are set up as Monitor Main outputs in the Monitor
panel can also be controlled by the standard Control Change Volume via MIDI channel 1.
With this, the main volume of the Fireface is controllable from nearly any MIDI equipped hardware device.
Even if you don't want to control all faders and pans, some buttons are highly desired to be
available in 'hardware'. These are mainly the Talkback and the Dim button, and the new monitoring options (listen to Phones submixes). Fortunately a Mackie Control compatible controller is
not required to control these buttons, as they are steered by simple Note On/Off commands on
MIDI channel 1.
The notes are (hex / decimal / keys):
Monitor Main: 3E / 62 / D 3
Dim: 5D / 93 / A 5
Mono: 2A / 42 / #F 1
Talkback: 5E / 94 / #A 5
An example of a small MIDI controller covering such MIDI functionality (and even some more) is
the Behringer BCN44. This little box has 4 pots and 8 buttons for all the above functions – for
less than 60 Euros.
Furthermore all faders of all three rows can be controlled via simple Control Change commands.
The format for the Control Change commands is:
Bx yy zz
x = MIDI channel
yy = control number
zz = value
The first row in TotalMix is addressed by MIDI channels 0 up to 3, the middle row by channels 4
up to 7 and the bottom row by channels 8 up to 11.
16 Controller numbers are used: 102 up to 117 (= hex 66 to 75).
With these 16 Controllers (= faders) and 4 MIDI channels each per row, up to 64 faders can be
controlled per row (as required by the HDSPe MADI).
: Sending MIDI strings might require the use of programmer's logic for the MIDI channel,
starting with 0 for channel 1 and ending with 15 for channel 16.
32.6 Loopback Detection
The Mackie Control protocol requires feedback of the received commands, back to the hardware controller. So usually TotalMix will be set up with both a MIDI input and MIDI output. Unfortunately any small error in wiring and setup will cause a MIDI feedback loop here, which then
completely blocks the computer (the CPU).
To prevent the computer from freezing, TotalMix sends a special MIDI note every 0.5 seconds
to its MIDI output. As soon as it detects this special note at the input, the MIDI functionality is
disabled. After fixing the loopback, check Enable MIDI Control under Options to reactivate the
TotalMix MIDI.
32.7 Stand-Alone MIDI Control
When not connected to a computer, the Fireface UC can be controlled directly via MIDI. To
unlock the special stand-alone MIDI control mode first activate MIDI control in TotalMix (En-able MIDI control). Turning this mode off is done with MIDI control deactivated.
Note
: When not needed the stand-alone MIDI operation should not be active, as the unit will
react on MIDI notes after power-on, and will also send MIDI notes.
Control is performed via the Mackie Control protocol. In stand-alone mode not all functions
known from TotalMix are available, because some of them aren't hardware, but software routines. Functions like Talkback, DIM, Mono, Solo, relative ganging of the faders, Monitor Main
and Monitor Phones are realized by complex software c ode, therefore not available in standalone MIDI control operation.
Still many functions, and especially the most important functions to control the Fireface UC, are
implemented in hardware, thus available also in stand-alone mode:
• All faders and pans of the first and third row
• Mute of the input signal per channel
• Ganging via 'Select'
• Choice of the routing destination, i.e. the current submix
• Sending of LED and display data to the MIDI controller
The second row (software playback) is skipped in stand-alone operation. The Fireface UC
sends display data as brief information, enabling an easy navigation through lines and rows.
Other data like PAN and miscellaneous status LEDs are supported as well.
In stand-alone mode the unit always operates in View Submix mode. Only this way the routing
destination can be changed, and several mixdowns/submixes can be set up quickly and easily.
Note
: After power-on the Fireface UC does not update the connected MIDI controller. Therefore
the controller’s faders will not be set to the values stored within the unit.
The stand-alone operation supports the following Mackie Control surface elements*:
*Tested with Behringer BCF2000 Firmware v1.07 in Mackie Control emulation for Steinberg mode.
Element: Meaning in Fireface:
Channel faders 1 – 8 volume
SEL(1-8) + DYNAMICS reset fader to Unity Gain
V-Pots 1 – 8 pan
pressing V-Pot knobs pan = center
CHANNEL LEFT or REWIND move one channel left
CHANNEL RIGHT or FAST FORWARD move one channel right
BANK LEFT or ARROW LEFT move eight channels left
BANK RIGHT or ARROW RIGHT move eight channels right
ARROW UP or Assignable1/PAGE+ move one row up
ARROW DOWN or Assignable2/PAGE- move one row down
Not all information to and around our products fit in a manual. Therefore RME offers a lot more
and detailed information in the Tech Infos. The very latest Tech Infos can be found on our
website, section News & Infos, or the directory \rmeaudio.web\techinfo on the RME Driver
CD. These are some of the currently available Tech Infos:
FireWire Audio by RME – Technical Background
FireWire 800 Hardware – Compatibility Problems
FireWire 800 under Windows XP SP2
Driver updates Fireface UC – Lists all changes of the driver updates.
SteadyClock: RME's new clock technology in theory and operation
DIGICheck: Analysis, tests and measurements with RME audio hardware
A description of DIGICheck, including technical background information.
HDSP System: TotalMix - Hardware and Technology
Background information on the digital mixer of the Hammerfall DSP/Fireface
Synchronization II (DIGI96 series)
Digital audio synchronization - technical background and pitfalls.
Installation problems - Problem descriptions and solutions.
ADI-8 Inside
Technical information about the RME ADI-8 (24-bit AD/DA converter).
HDSP System: Notebook Basics - Notebook Hardware
HDSP System: Notebook Basics - The Audio Notebook in Practice
HDSP System: Notebook Basics - Background Knowledge and Tuning
HDSP System: Notebook Tests - Compatibility and Performance
Many background information on laptops. Tests of notebooks.
Digital signals consist of a carrier and the data. If a digital signal is applied to an input, the receiver has to synchronize to the carrier clock in order to read the data correctly. To achieve this,
the receiver uses a PLL (Phase Locked Loop). As soon as the receiver meets the exact frequency of the incoming signal, it is locked. This Lock state remains even with small changes of
the frequency, because the PLL tracks the receiver's frequency.
If an ADAT or SPDIF signal is applied to the Fireface UC, the corresponding input LED starts
flashing. The unit indicates LOCK, i. e. a valid input signal (in case the signal is also in sync, the
LED is constantly lit, see below).
Unfortunately, LOCK does not necessarily mean that the received signal is correct with respect
to the clock which processes the read out of the embedded data. Example [1]: The Fireface is
set to 44.1 kHz internally (clock mode Master), and a mixing desk with ADAT output is connected to input ADAT. The corresponding LED will show LOCK immediately, but usually the
mixing desk's sample rate is generated internally (also Master), and thus slightly higher or lower
than the Fireface's internal sample rate. Result: When reading out the data, there will frequently
be read errors that cause clicks and drop outs.
Also when using multiple inputs, a simple LOCK is not sufficient. The above described problem
can be solved elegantly by setting the Fireface from Master to AutoSync (its internal clock will
then be the clock delivered by the mixing desk). But in case another, un-synchronous device is
connected, there will again be a slight difference in the sample rate, and therefore clicks and
drop outs.
In order to display those problems optically at the device, the Fireface includes SyncCheck
checks all clocks used for synchronicity. If they are not synchronous to each other (i. e. absolutely identical), the SYNC LED of the asynchronous input flashes. In case they are completely
synchronous, all LEDs are constantly lit. In example 1 it would have been obvious that the LED
ADAT kept on flashing after connecting the mixing desk.
In practice, SyncCheck allows for a quick overview of the correct configuration of all digital devices. So one of the most difficult and error-prone topics of the digital studio world finally becomes easy to handle.
The same information is presented in the Fireface's Settings dialog. In the status display Input State the state of all clocks is decoded and shown as simple text (No Lock, Lock, Sync).
The term Zero Latency Monitoring has been introduced by RME in 1998 for the DIGI96 series
of audio cards. It stands for the ability to pass-through the computer's input signal at the interface directly to the output. Since then, the idea behind has become one of the most important
features of modern hard disk recording. In the year 2000, RME published two ground-breaking
Tech Infos on the topics Low Latency Background, which are still up-to-date: Monitoring, ZLM and ASIO, and Buffer and Latency Jitter, both found on the RME Driver CD and the RME website.
How much Zero is Zero?
From a technical view there is no zero. Even the analog pass-through is subject to phase errors, equalling a delay between input and output. However, delays below certain values can
subjectively be claimed to be a zero-latency. This applies to analog routing and mixing, and in
our opinion also to RME's Zero Latency Monitoring. The term describes the digital path of the
audio data from the input of the interface to its output. The digital receiver of the Fireface UC
can't operate un-buffered, and together with TotalMix and the output via the transmitter, it
causes a typical delay of 3 samples. At 44.1 kHz this equals about 68 µs (0.000068 s), at 192
kHz only 15 µs. The delay is valid for ADAT and SPDIF in the same way.
Oversampling
While the delays of digital interfaces can be disregarded altogether, the analog inputs and outputs do cause a significant delay. Modern converter chips operate with 64 or 128 times oversampling plus digital filtering, in order to move the error-prone analog filters away from the audible frequency range as far as possible. This typically generates a delay of one millisecond. A
playback and re-record of the same signal via DA and AD (loopback) then causes an offset of
the newly recorded track of about 2 ms. The exact delays of the Fireface UC are:
Sample frequency kHz 44.1 48 88.2 96 176.4 192
AD (43.2 x 1/fs) ms 0.98 0.9 0.49 0.45
AD (38.2 x 1/fs) ms 0.22 0.2
DA (28 x 1/fs) ms *
Buffer Size (Latency)Windows: This option found in the Settings dialog defines the size of the buffers for the audio
data used in ASIO and WDM (see chapter 13).
Mac OS X: The buffer size is defined within the application. Only some do not offer any setting.
For example iTunes is fixed to 512 samples.
General: A setting of 64 samples at 44.1 kHz causes a latency of 1.5 ms, for record and playback each. But when performing a digital loopback test no latency/offset can be detected. The
reason is that the software naturally knows the size of the buffers, therefore is able to position
the newly recorded data at a place equalling a latency-free system.
AD/DA Offset under ASIO and OS X: ASIO (Windows) and Core Audio (Mac OS X) allow for the
signalling of an offset value to correct buffer independent delays, like AD- and DA-conversion or
the Safety Buffer described below. An analog loopback test will then show no offset, because
the application shifts the recorded data accordingly. Because in real world operation analog
record and playback is unavoidable, the drivers include an offset value matching the Fireface's
converter delays.
Therefore, in a digital loopback test a negative offset of about 3 ms occurs. This is no real
problem, because this way of working is more than rare, and usually the offset can be compensated manually within the application. Additionally, keep in mind that even when using the digital
I/Os usually at some place an AD- and DA-conversion is involved (no sound without...).
Note
: Cubase and Nuendo display the latency values signalled from the driver separately for
record and playback. While with our former cards these values equalled exactly the buffer size
(for example 3 ms at 128 samples), the Fireface displays an additional millisecond – the time
needed for the AD/DA-conversion. Playback even shows another millisecond added – see
Safety Buffer.
Safety Buffer
An additional small Safety Buffer on the playback side only has proven to be very efficient and
useful. The Fireface UC uses a fixed additional buffer of 32 samples, which is added to the current buffer size. The main advantage is the ability to use lowest latency at highest CPU loads.
Furthermore, the fixed buffer does not add to the latency jitter (see Tech Info), the subjective
timing is extraordinary.
Core Audio's Safety Offset
Under OS X, every audio interface has to use a so called Safety Offset, otherwise Core Audio
won't operate click-free. The Fireface UC uses a safety offset of 28 samples. This offset is signalled to the system, and the software can calculate and display the total latency of buffer size
plus AD/DA offset plus Safety Offset for the current sample rate.
35.3 USB Audio
USB audio is in several ways different from PCI based audio interfaces.
A Fireface UC can achieve a performance similar to a PCI or PCI Express card when used with
an optimal PC. Low CPU load and click-free operation even at 48 samples buffer size are indeed possible on current computers. However, using older computers a simple stereo playback
will already cause a CPU load of more than 30%.
A computer blocked for a short time – no matter if ASIO or WDM – will loose one or more data
packets. Such problems can only be solved by increasing the buffer size (and with this the latency).
The Fireface UC features a unique data checking,
detecting errors during transmission via USB and
displaying them in the Settings dialog. Additionally the
Fireface provides a special mechanism to continue
recording and playback in spite of drop-outs, and to correct
the sample position in real-time.
Detailed information on this topic can be found in the Tech Info USB Audio by RME – Technical Background on our website:
When activating the Double Speed mode the Fireface UC 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 UC 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
ADAT
3/4
ADAT
5/6
ADAT
7/8
ADAT
- - -
As the transmission of double rate signals is done at standard sample rate (Single Speed), the
ADAT output still delivers 44.1 kHz or 48 kHz.
35.5 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 results in only two channels per
optical output. There are few devices using this method.
In Quad Speed mode the Fireface UC automatically uses Sample Multiplexing. One channel's
data is distributed to four channels according to the following table:
Analog In 1 2 3 4 5 6 7 8
DS Signal
Port
1/2/3/4
ADAT
As the transmission of quad rate signals is done at standard sample rate (Single Speed), the
ADAT output still delivers 44.1 kHz or 48 kHz.
The SPDIF (AES) output of the Fireface UC provides 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 Un-balanced
Impedance 110 Ohm 75 Ohm
Level 0.2 V up to 5 Vss 0.2 V up to 0.5 Vss
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 necessary).
The outstanding signal to noise ratio of the Fireface's AD-converters can be verified even without expensive test equipment, by using record level meters of various software. But when activating the DS and QS mode, the displayed noise level will rise from -109 dB to -104 dB at 96
kHz, and –82 dB at 192 kHz. This is not a failure. The software measures the noise of the whole
frequency range, at 96 kHz from 0 Hz to 48 kHz (RMS unweighted), at 192 kHz from 0 Hz to 96
kHz.
When limiting the measurement range from 20 Hz to 20 kHz (so called audio bandpass) the
value would be -110 dB again. This can be verified with RME's DIGICheck. The function Bit Statistic & Noise measures the noise floor by Limited Bandwidth, ignoring DC and ultrasound.
The reason for this behaviour is the noise shaping technology of the analog to digital converters. They move all noise and distortion to the in-audible higher frequency range, above 24 kHz.
That’s how they achieve their outstanding performance and sonic clarity. Therefore the noise is
slightly increased in the ultrasound area. High-frequent noise has a high energy. Add the doubled (quadrupled) bandwidth, and a wideband measurement will show a significant drop in
SNR, while the human ear will notice absolutely no change in the audible noise floor.
35.8 SteadyClock
The SteadyClock technology of the Fireface UC guarantees an excellent performance in all
clock modes. Thanks to a highly efficient jitter suppression, the AD- and DA-conversion always
operates on highest sonic level, being completely independent from the quality of the incoming
clock signal.
SteadyClock has been originally 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).
Using the Fireface's input signals
SPDIF and ADAT, 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.
Common interface jitter values in real
world applications are below 10 ns, a
very good value is less than 2 ns.
The screenshot shows an extremely jittery SPDIF signal of about 50 ns jitter (top graph, yellow).
SteadyClock turns this signal into a clock with less than 2 ns jitter (lower graph, blue). The signal processed by SteadyClock is of course not only used internally, but also used to clock the
digital outputs. Therefore the refreshed and jitter-cleaned signal can be used as reference clock
without hesitation.
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 output 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.
XLR jacks of analog inputs
The XLR jacks are wired according to international standards:
1 = GND (shield)
2 = + (hot)
3 = - (cold)
TRS Phones jack
The analog monitor output on
the front is accessible through
a stereo ¼" TRS jack. This
allows a direct connection of
headphones. In case the output
should operate as Line output,
an adapter TRS plug to RCA
phono plugs, or TRS plug to TS
plugs is required.
The pin assignment follows
international standards. The left
channel is connected to the tip,
the right channel to the ring of
the TRS jack/plug.