After the driver installation (see chapter 10 / 19) connect the TRS-jacks with the analog signal source. Try to achieve an optimum input level by adjusting the source itself. Raise the sources output level until the peak level meters in TotalMix reach about 3 dB. In case the levels do not match at all, the input sensitivity can also be changed via internal jumpers (see chapter 35). The analog line inputs of the Multiface can be used with +4 dBu and -10 dBV signals. The electronic input stage can handle balanced (TRS jacks) and unbalanced (TS jacks) input signals correctly. The Multiface's digital outputs provide SPDIF (AES/EBU compatible) and ADAT optical signals at the corresponding ports. An additional stereo output is available on the front. This output is a very low impedance type, which can also be used to connect headphones.
6.3 Notes on Laptops and CardBus
The HDSP system uses the notebooks PCMCIA type II port as CardBus interface. Compared to a PC-Card, which only has access to the outdated ISA-bus, CardBus is a 32 bit PCI interface. Like with a desktop system its not possible to remove a PCI device while in operation. First the operating system has to receive a removal request, then the device has to be stopped. Finally the card can be pulled out of the PCMCIA slot. Windows When inserting the CardBus card it usually will be detected automatically by the notebook hardware and then the operating system. A beep signals the detection. In rare cases detection will fail. If so, simply remove the card and insert it again. To remove the hardware click on the green arrow symbol in the systray. It is possible to stop the HDSP directly, or to first call up the info dialog by double clicking the symbol, and then stopping it. Mac OS X When inserting the CardBus card it usually is detected automatically by the notebook hardware and then by the Mac OS. A CardBus icon will appear on the top menu. A mouse click on the icon opens a drop-down menu, showing the card's name (Hammerfall DSP) and an option to switch it off. To remove the CardBus card click on the menu entry 'Power off card'. The Mac OS internally de-installs the CardBus card and switches off power (the red Host LED begins to blink). The card can now be pulled out of the PCMCIA slot.
Tech Infos The Hammerfall DSP System was tested thoroughly on several notebooks by RME. Basically the CardBus card operates with any tested device. Our website hosts several interesting Tech Infos with background information: HDSP-System: Notebook Tests Compatibility and Performance HDSP System: Notebook Basics - Notebook Hardware HDSP System: Notebook Basics - Background Knowledge and Tuning

The mobile operation of the HDSP system can cause problems. Explanations and solutions on digital noise, ground loops, headphone operation and Line Out wiring, power supplies and the mobile operation with battery can be found in the Tech Info: HDSP System: Notebook Basics - The Audio Notebook in Practise.

7. Accessories

RME offers several optional accessories. Additionally parts of the HDSP system are available separately. Part Number 36000 Description 19, 1UH Universal rack holder
This 19" rack holder has holes for Digiface and Multiface. Two units can be installed side by side in any combination. The rack holder also includes holes for nearly all 19" half-rack units from other manufacturers. Firewire cable IEEE1394 6M/6M, 1 m (3.3 ft) Firewire cable IEEE1394 6M/6M, 3 m (9.9 ft) Firewire cable IEEE1394 6M/6M, 5 m (16.4 ft) Firewire cable IEEE1394 6M/6M, 10 m (32.8 ft)
Firewire cable for the HDSP system, both sides 6-pin male. Cable longer than 16 ft is not allowed for FireWire, therefore hard to get in computer shops. However the HDSP system does not use FireWire protocol, therefore can operate flawlessly even with a cable length of up to 50ft (15 m). Optical cable, TOSLINK, 0.5 m (1.6 ft) Optical cable, TOSLINK, 1 m (3.3 ft) Optical cable, TOSLINK, 2 m (6.6 ft) Optical cable, TOSLINK, 3 m (9.9 ft) Optical cable, TOSLINK, 5 m (16.4 ft) Optical cable, TOSLINK, 10 m (33 ft)
Standard lightpipe with TOSLINK connectors, RME approved quality. 37011 Power supply for HDSP CardBus card
Robust and light weigth switching power supply, 100V-240V AC, 12V 1.25 A DC.

8. Warranty

Each individual Hammerfall DSP undergoes comprehensive quality control and a complete test at RME before shipping. The usage of high grade components allow us to offer a full two year warranty. We accept a copy of the sales receipt as valid warranty legitimation. If you suspect that your product is faulty, please contact your local retailer. The warranty does not cover damage caused by improper installation or maltreatment - replacement or repair in such cases can only be carried out at the owners expense. RME does not accept claims for damages of any kind, especially consequential damage. Liability is limited to the value of the Hammerfall DSP. The general terms of business drawn up by Synthax Audio AG apply at all times.

9. Appendix

RME news, driver updates and further product information are available on our website: http://www.rme-audio.com If you prefer to read the information off-line, you can browse through a complete copy of the RME website, found on the RME Driver CD (in the \rmeaudio.web directory). Manufacturer: IMM Elektronik GmbH, Leipziger Strasse 32, D-09648 Mittweida

I/O Box State This field displays the current state of the I/O-box. Error: Detected: Connected: Disconnected: I/O-box not connected or missing power The interface has found an I/O-box and tries to load the firmware Communication between interface and I/O-box operates correctly Communication between interface and I/O-box has been interrupted, I/Obox continues operation
Word Clock Out 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 sample rate, the output word clock is 48 kHz. Clock Mode The unit can be configured to use its internal clock source (Master), or the clock source predefined via Pref. Sync Ref (AutoSync). Pref. Sync Ref. Used to pre-select the desired clock source. If the selected source isn't available, the unit will change to the next available one. The current clock source and sample rate is displayed in the AutoSync Ref display. The automatic clock selection checks and changes between the clock sources Word Clock, ADAT, ADAT Sync and SPDIF. System Clock Shows the current clock state of the HDSP system. The system is either Master (using its own clock) or Slave (see AutoSync Ref).
11.2 Clock Modes - Synchronisation
In the digital world, all devices are either the Master (clock source) or a Slave synchronized to the master. Whenever several devices are linked within a system, there must always be a single master clock. The Hammerfall DSPs intelligent clock control is very user-friendly, being able to switch between clock modes automatically. Selecting AutoSync will activate this mode. In AutoSync mode, the system constantly scans all digital inputs for a valid signal. If this signal corresponds with the current playback sample rate, the card switches from the internal quartz (AutoSync Ref displays 'Master') to a clock generated from the input signal (AutoSync Ref displays 'Slave'). This allows on-the-fly recording, even during playback, without having to synchronize the card to the input signal first. It also allows immediate playback at any sample rate without having to reconfigure the card. AutoSync guarantees that normal 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 Hammerfall DSP, AutoSync may cause feedback in the digital carrier, so synchronization breaks down. To remedy this, switch the HDSPs clock mode over to 'Master'. Remember that a digital system can only have one master! If the HDSPs clock mode is set to 'Master', all other devices must be set to Slave.

The newest information can always be found on our website www.rme-audio.com, section FAQ, Latest Additions. The input signal cannot be monitored in real-time ASIO Direct Monitoring has not been enabled, and/or monitoring has been disabled globally. The 8 ADAT channels dont seem to work The optical output has been switched to SPDIF. The ADAT playback devices are still usable by routing and mixing them in TotalMix to other outputs. Low Latency ASIO operation under Windows 2000/XP on single CPU systems: To use ASIO at lowest latencies under Windows 2000/XP even when only having one CPU, the system performance has to be optimized for background tasks. Go to >Control Panel/ System/ Advanced/ Performance Options<. Change the default 'Applications' to 'Background tasks'. The lowest usable latency will drop from 23 ms to around 3 ms.
Playback works, but record doesnt 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 HDSP system 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 systemss 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 systems clock mode to Master. Increase the buffer size of the hard disk cache. Activate Busmaster mode for the hard disks. In case of a recently done BIOS update of the motherboard: Propably 'Load BIOS Defaults' was loaded instead of 'Load Setup Defaults'. This sets the 'PCI Latency Timer' to 0 (default: 32). ADAT timecode is running, but Cubase does not start 'Play' automatically The input displayed in Sync Ref is not in sync mode. Sync mode is essential, because ADATs so-called time code is really a sample position, and is therefore only valid for synchronous audio data. Sync is displayed (referring to the cards clock), but the incoming data is not in sync with the sample position received at the ADAT Sync In. Then Cubase does not start. Remedy: Set Pref. Sync Ref to the input corresponding to the received ADAT Sync signal. Sync mode wasn't activated (button in the transport panel), or ASIO 2.0 has not been chosen as the SMPTE sync source. The ADAT timecode is not in sync The tape is formatted to 48 kHz, but played back at 44.1 kHz (Pitch). This 'Blackface' problem cannot be solved in a satisfactory way.

17.2 Installation

Hammerfall DSP is found in the Device Manager (>Settings/ Control Panel/ System<), category 'Sound-, Video- and Gamecontroller'. A double click on ' Hammerfall DSP ' starts the properties dialog. Choosing 'Resources' shows Interrupt and Memory Range. The newest information on hardware problems can always be found on our website www.rmeaudio.com, section FAQ, Hardware Alert: about incompatible hardware. The dialog 'New hardware component found does not appear: Check whether the CardBus card is completely inserted into the PCMCIA slot, or the PCI interface is correctly inserted in the PCI slot. The card and drivers have been installed correctly, but playback does not work: Check whether the Hammerfall DSP appears in the Device Manager. If the ' Hammerfall DSP device has a yellow exclamation mark, then there is an address or interrupt conflict. Even if there is no yellow exclamation mark, it's still worth checking the Resources tab.

All other PCI cards, and CardBus with 15-pin flat connector When the update fails (status: failure) the flash process should be repeated several times, until no error message occurs anymore. If the failure message is displayed nonetheless, the interface will most propably no longer work when the computer is switched off and on again. The interface then has to be re-programmed at the factory. We have invested a lot of work to prevent the system from getting in this state. If it happens despite our efforts, the best advice we can give is to not switch off the computer! As long as it is not switched off the old programming of the PCI/CardBus interface will stay active, and you can continue to work with the system using the old drivers
20. Configuring the Multiface

20.1 Settings Dialog

Configuring the Multiface is done via its own settings dialog. The panel 'Settings' can be opened by clicking on the hammer icon in the dock. The mixer of the Hammerfall DSP System, TotalMix, can be opened by clicking on the mixer icon in the dock. The Hammerfall DSPs 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: Input selection Output mode Output channel status Synchronization behaviour Input and output status display
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 the user precise information about the current status of the system, and the status of all digital signals. SyncCheck indicates whether there is a valid signal (Lock, No Lock) for each input (Word Clock, ADAT, SPDIF), or if there is a valid and synchronous signal (Sync). The AutoSync Reference display shows the input and frequency of the current sync source.
SPDIF In Defines the input for the SPDIF signal. 'Coaxial' relates to the RCA socket, 'Optical' to the optical TOSLINK input. SPDIF Out The SPDIF output signal is constantly available at the phono plug. After selecting 'Optical' it is also routed to the optical TOSLINK output. For further details about the settings Professional, Emphasis and Non-Audio, please refer to chapter 26.2. SPDIF Freq. Displays the sample rate of the signal at the SPDIF input. I/O Box Disconnect interrupts the communication between I/O-box and PCI or CardBus card. In case the Multiface has been configured using the Settings dialog and TotalMix, Disconnect allows to use it Stand-Alone (without a connected computer), after a power supply has been attached. The status display below shows the current state of the I/O-boxt: Error: Detected: Connected: Disconnected: I/O-box not connected or missing power The interface has found an I/O-box and tries to load the firmware Communication between interface and I/O-box operates correctly Communication between interface and I/O-box has been interrupted, I/Obox continues operation

26.2 SPDIF

Input The SPDIF input is configured in the Settings dialog, available by a click on the hammer symbol in the Task Bar's system tray. The HDSP system accepts all commonly used digital sources as well as SPDIF and AES/EBU. Channel status and copy protection are ignored. 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 only connected to pin 1 of the XLR - not to the phono plug.
The ground-free design, with transformers for coaxial digital inputs and outputs, offers a trouble-free connection of all devices along with perfect hum rejection and full AES/EBU compatibility. Output In SPDIF mode, identical signals are available at both the optical and the coaxial output. An obvious use for this would be to connect two devices, i.e. using the HDSP as a splitter (distribution 1 on 2). 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 HDSP system ignores the received header and creates a totally new one for its output signal. Note that in record or monitor modes, set emphasis bits will disappear. Recordings originally done with emphasis should always be played back with the emphasis bit set! This can be done by selecting the Emphasis switch in the Settings dialog (field SPDIF Out). This setting is updated immediately, even during playback. Note: Recordings with (pre-) emphasis show a treble boost (50/15 s), which has to be compensated at playback. Therefore, when selecting Emphasis all analog outputs will be processed by a treble filter based on 50/15s, which sounds like a high cut.
The Multifaces new output header is optimized for largest compatibility with other digital devices: 32 kHz, 44.1 kHz, 48 kHz, 88.2 kHz, 96 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 s Aux bits Audio Use
Professional AES/EBU equipment can be connected to the Multiface thanks to the transformerbalanced coaxial output, and the Professional format option with 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! The audio bit in the header can be set to 'Non-Audio'. This is often necessary when Dolby AC-3 encoded data is sent to external decoders (surround-sound receivers, television sets etc. with AC-3 digital inputs), as these decoders would otherwise not recognize the data as AC-3.

28. TotalMix: Routing and Monitoring

28.1 Overview

The Multiface includes a powerful digital real-time mixer, the Hammerfall DSP mixer, based on RMEs 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 Multiface allows for up to 10 (!) fully independent stereo submixes. On an analog mixing desk, this would equal 20 (!) 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 multiclient 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 (hardware output is Peak only). These level displays are very useful to determine the precense 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), the record signal usually stays un-altered. TotalMix 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 hardwares outputs are connected post-fader, thus displaying the actual output level.

28.2 The User Interface

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 Multiface provides 18 input channels, 18 software playback channels, and 20 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 independent. 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.
28.3 Elements of a Channel
A single channel consists of various elements: Input channels and playback channels each have a mute and solo button. Below there is the panpot, 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 white area shows the channel name. Selecting one or more channels is done by clicking on the white label which turns orange then. A click 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.

28.4 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. 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 AN 3/4 and the faders above are the only active ones, same for AN5/6 and so on. Back to AN1/2. 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 AN1/2. 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.

You see, it is very easy to set up a specific submix for whatever output: select output channel, set up 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), labeled 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 7+8'. The list disappears, the routing field no longer shows 'AN1+2', but ' AN 7+8'. 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 7, you can see the level of what you are hearing from output 7. 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 7 and 8. 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 8 as well. In short: While editing the Submix AN7/AN8 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 in 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 7+8'. Now, Out 4 has already been set to 'AN 7+8' 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 three checkmarks, at 'AN 3+4', 'AN 7+8' and 'Phones'. 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.

29. TotalMix: The Matrix

29.1 Overview
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 HDSP looks and works like a conventional patchbay, adding functionality way beyond comparable hardware and software soutions. 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.
29.2 Elements of the Matrix View
The visual design of the TotalMix Matrix is mainly determined by the architecture of the HDSP system: 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. To maintain overview when the window size has been reduced, the left and upper labels are floating. They won't left the visible area when scrolling.

29.3 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. Moving a fader in row 1 or 2 in TotalMix view, only the specific levels (max. 2) of this routing will change within the Matrix. But moving a fader in row 3 will make all vertically activated levels move at once (for example 19/20, Phones output). A gain field marked orange indicates activated mute status. Mute can only be changed in the mixer view.

30.4 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 TotalMix, so be careful with this function!
30.5 Recording a Subgroup (Loopback)
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 of 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 Ctrlkey. 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.
* Note: Because of a technical limitation the input's level meter no longer shows the input signal of the hardware which still can be routed by TotalMix but the loopback signal. This gives the false impression of the loopback signal being present at the TotalMix mixer input, which is not the case.
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 rivaled 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 only allowed as long as the monitoring is routed in both software and TotalMix to a different channel than the active subgroup recording one.
Recording a Software's playback 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 in only one track. For example when using two microphones when 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.

30.6 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 headphones (outputs 19/20). A direct routing In 1 to Out 19/20 for monitoring had been set up already. The external reverb is connected to a free output, for example channel 8. In active mode Submix View click on channel 8 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 19/20 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 8 in the bottom row again and drag 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 labeled 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 a 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 19/20. 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 30.5, 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.

33.7 General

Power supply: external switching power supply, 100 - 240 V AC, 15 Watt Current at 12 Volt operating voltage, unloaded: 500 mA (6 Watt) Current at 12 Volt operating voltage, loaded: 760 mA (9 Watt) Typical power consumption: 9 Watt Voltage range: DC 8 V 28 V, AC 8 V 20 V Dimensions (WxHxD): 215 x 44 x 115 mm (8.5" x 1.73" x 4.5") Weight: 1.5 kg ( 3.3 lbs) Temperature range: +5 up to +50 Celsius (41 F up to 122F) Relative humidity: < 75%, non condensing

34. Technical Background

34.1 Lock and SyncCheck
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 Multiface, 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 Multiface is set to 44.1 kHz internally (clock mode Master), and a mixing desk with ADAT output is connected to input ADAT1. 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 Multiface'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 Multiface 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 Multiface includes SyncCheck. It 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 1 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 Multiface's Settings dialog. In the status display SyncCheck the state of all clocks is decoded and shown as simple text (No Lock, Lock, Sync).