Thank you for choosing the Hammerfall DSP MADI. This unique audio system is capable of
transferring digital audio data directly into a computer, from any device equipped with a MADI
interface. Installation is simple, even for the inexperienced user, thanks to the latest Plug and
Play technology. The numerous unique features and well thought-out configuration dialog puts
the Hammerfall DSP MADI at the very top of the range of digital audio interface cards.
The package includes drivers for Windows 2000/XP. An ALSA driver for Linux is under
development (see chapter 7.5).
Our high-performance philosophy guarantees maximum system performance by executing all
functions directly in hardware and not in the driver (i.e. the CPU).
2. Package Contents
Please check that your Hammerfall DSP MADI package contains each of the following:
• HDSP MADI PCI card
• HDSP MADI expansion board
• Quick Info guide
• RME Driver CD
• MIDI breakout cable
• Expansion Board ribbon cable (14-conductor)
3. System Requirements
• Windows 2000/XP, Linux
• PCI Interface: a free PCI rev. 2.1 Busmaster slot
Note: Examples and detailed descriptions of suitable audio desktop systems can be found in
the Tech Info RME Reference PCs: Hardware recommendations.
4. Brief Description and Characteristics
• Hammerfall design: 0% (zero!) CPU load, even using all 128 ASIO channels
• 24 or 32 bit, 4 byte (stereo 8 byte)
This format is compatible with 16-bit and 20-bit. Resolutions below 24-bit are handled by the
audio application.
MME:
• 16 bit, 2 byte (stereo 4 byte)
• 20 bit, 3 byte MSB (stereo 6 byte)
• 20 bit, 4 byte MSB (stereo 8 byte)
• 24 bit, 3 byte (stereo 6 byte)
• 24 bit, 4 byte MSB (stereo 8 byte)
• 32 bit, 4 byte (stereo 8 byte)
The card supports multi-device operation only, channel interleave operation is not supported.
6. Hardware Installation
Before installing the PCI card, please make sure the computer is switched off and the
power cable is disconnected from the mains supply. Inserting or removing a PCI card while
the computer is in operation will cause irreparable damage to both motherboard and card!
1. Disconnect the power cord and all other cables from the computer.
2. Remove the computer's housing. Further information on how to do this can be obtained
from your computer´s instruction manual.
3. Important: Before removing the HDSP MADI from its protective bag, discharge any static in
your body by touching the metal chassis of the PC.
4. Connect the HDSP MADI card with the Expansion Board using the supplied flat ribbon
cable. Note: The connectors on the cable cannot be plugged in the wrong way round.
5. Insert the HDSP MADI firmly into a free PCI slot, press and fasten the screw.
6. Insert the Expansion Board and fasten the screw.
7. Replace the computer's housing.
8. Reconnect all cables including the power cord.
Note: If neither word clock I/O nor MIDI I/O is required, it is not necessary to install the
Expansion Board at all (i.e. leave out steps 4 and 6).
After the PCI card has been installed correctly (see 6. Hardware Installation), and the computer
has been switched on, Windows will recognize the new hardware component and start its
‘Hardware Wizard’. 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
\HDSP_MADI_w2k on the RME Driver CD.
Windows will install the Hammerfall DSP System driver, and will register the card in the system
as a new audio device. After a reboot the HDSP MADI is ready for use.
HDSP MADI can be easily configured using the HDSP Settings dialog (see section 10.1)
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 installation of the card,
do not attempt to install the drivers manually! An installation of drivers for non-recognized
hardware will cause a blue screen when booting Windows!
7.2 Driver Update
RME's driver updates often include a new madi.inf file. Also the revision number of the
hardware might change (after a flash update). To prevent Windows 2000/XP from using an old
madi.inf, or to copy some of the old driver files, be sure NOT to let Windows search for the
driver! Instead tell Windows what to do.
Under Control Panel /System /Device Manager /Sound, Video and Game Controllers /RME
Hammerfall DSP MADI/Properties /Driver you'll find the 'Update Driver' button. 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.
7.3 Flash Update
The Flash Update Tool updates the HDSP MADI's hardware to the latest version. It requires an
already installed driver.
Start the program fut_madi_xxx.exe. The Flash Update Tool displays the current version of the
HDSP MADI (200 or up), and whether it needs an update or not. If so, then simply press the
'Update' button. A progress bar shows how several actions are performed. When the flash
update process is finished, 'Success' will be displayed.
If more than one interface cards are installed, all cards can be flashed by changing to the next
tab and repeating the process.
After the update the PCI card needs to be resettet. This is done by powering down and shutting
off the PC. A warm boot is not enough.
When the update fails (status: failure), the card's second BIOS will be used from the next cold
boot on (Secure BIOS Technology). Therefore the card stays fully functional. The flash process
should then be tried again on a different computer.
Note for Windows 2000/XP users: Because of the changed hardware revision, Windows
2000/XP will start the hardware assistant and wants to install new drivers. Do NOT let Windows
search for new drivers, but follow the instructions given in chapter 7.2 and manually perform a
driver update.
A deinstallation of the HDSP's 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 additonal autorun entries of
TotalMix, the Settings dialog, and the registering of the ASIO driver. Those entries can be
removed from the registry through a software deinstallation request. This request can be found
(like all deinstallation entries) in Control Panel, Software. Click on the entry 'RME Hammerfall
DSP MADI'.
7.5 Linux/Unix
An ALSA driver for Linux/Unix is already under development and should be available soon.
Even TotalMix has been ported to Linux. Further information on ALSA is available at
HDSP MADI consists of the main PCI board and an Expansion Board. All the essential
electronics are located on the PCI card, so it will also work without the Expansion Board.
The main board's bracket
has two MADI interfaces,
optical and coaxial input
and output each, a MADI
erro LED and the analog
line/headphone output.
Configuration of inputs
and outputs is done via
the Settings dialog
(started by clicking on the
hammer symbol in the system tray).
Identical signals are available at both the optical and the coaxial output. An obvious use for this
would be to simply connect two devices, i.e. using the HDSP MADI as a splitter.
The Expansion Board's
bracket has the word clock
input and output. Next to
the input BNC socket, a
green LED displays the
word clock input's lock
status. Between the BNC
sockets, word clock
termination can be
activated and verified by a
yellow LED.
The included breakout cable is connected to the 9-pin Mini-DIN connector and provides two
MIDI inputs and outputs.
8.2 MADI I/Os
The BNC input's ground-free design is built according to AES10-1991. The input impedance is
75 Ohm. It will operate error-free from about 180 mVpp.
The optical input and output uses a FDDI (ISO/IEC 9413-3) compatible optical module,
according to AES10-1991. More information can be found in chapter 18, MADI Basics.
HDSP MADI includes automatic input selection (Safe Mode Input). In case the current input
signal fails, the unit switches to the other input immediately. This mode, called redundancy
mode, offers improved safety against errors on the transmission line. Switching the inputs is
done in about one second. Redundancy operation is disaplayed in the Settings dialog.
The BNC output is built according to AES10-1991. The output's impedance is 75 Ohm. The
output voltage will be 600 mVpp when terminated with 75 Ohm. Changing the internal blue
jumper X4 to the upper position, the output voltage is increased to 1.2 Vpp. This setting is not
intended to be used in normal operation. But in case of a very long or 'lossy' coaxial cable, this
setting may ensure an error-free operation of the transmission line.
HDSP MADI includes SteadyClock, guaranteeing an excellent performance in all clock modes.
Its highly efficient jitter suppression refreshes and cleans up any clock signal, and provides it as
reference clock at the BNC output (see section 19).
Input
The transformer-isolated word clock input is loacated on the Expansion Board. It is activated
via Pref. Sync Ref / Wordclock. As soon as a valid word clock signal is detected, the green
'Lock' LED beside the input jack lights up, and in the Settings dialog the field AutoSync Ref
shows Word. Below the detected frequency of the word clock is shown.
The word clock input is shipped as high impedance type (not terminated). A push switch allows
to activate internal termination (75 Ohms). The switch is found between the BNC jacks. Use a
small pencil or similar and carefully push the blue switch so that it snaps into its lock position.
When termination is active the yellow TERM LED will be lit. Another push will release it again
and de-activate the termination.
Due to the HDSP MADI's outstanding clock control a synchronization of the output signal to the
input signal is not only possible at identical sample rates, but also at half, quarter, double and
quad sample rates.
Example 1: A playback or recording at 44.1 kHz can be synchronized via an external signal of
44.1 kHz, 88.2 kHz or 176.4 kHz.
Example 2: A playback or recording at 192 kHz can be synchronized via an external signal of
48 kHz, 96 kHz or 192 kHz.
The input accepts all those frequencies fully automatically.
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.
Output
The word clock output of the HDSP MADI is always active. It provides the current sample
frequency of the HDSP MADI as word clock signal. As long as the HDSP MADI operates in
'Master' mode (field 'Clock Mode'), the word clock will be fixed to the current sample rate. In
'AutoSync' mode the sample rate is identical to the one present at the currently chosen input
(Pref. Sync Ref). Without a valid input signal, the card will change between the inputs
automatically. As long as no valid input signal is found, the card will stay in Master mode. This
way, the card will always generate a valid output signal.
The word clock signal received by the HDSP MADI can be distributed to other devices by using
the word clock output. With this the usual T-adapter can be avoided, and the HDSP MADI
operates as Signal Refresher. 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
Adaptation Circuit) plus SteadyClock guarantee a secure function also with most critical
word clock signals.
HDSP MADI offers a hi-quality analog monitor output. The short circuit protected stereo line
output provides high output level, low impedance, and is available via a 6.3 mm (1/4") TRS
jack. Therefore it is also suitable for a direct use with headphones.
The analog output is directly driven from the channels 63/64. Its output volume is controlled by
the hardware output faders of channel 63/64 in TotalMix. Additionally the analog output can
play back any input or playback signal (submix, for example Preset 5, 6 and 7).
RME's unique Speaker Protection reduces noise when switching the computer on and off, so
there is no problem even when using active monitors.
In case the output should operate as
line out, 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.
8.5 MIDI
Hammerfall DSP MADI offers two MIDI I/Os via 5-pin DIN jacks. The MIDI ports are added to
the system by the driver. Using MIDI capable software, these ports can be accessed as MADI
MIDI In 1 (1), MADI MIDI In 2 (2), MADI MIDI Out 1 (1) and MADI MIDI Out 2 (2). The brackets
include the card number.
The MIDI inputs can not operate multiclient, so a MIDI input signal can not be distributed to
several programs at the same time. Such a functionality can be achieved with third party tools.
The third MIDI port, MADI MIDI In 3 (1) and MADI MIDI Out 3 (1), receives and transmits MIDI
data via MADI. This allows for a direct communication between systems with HDSP MADI
cards. Additionally MIDI data can be transmitted from/to RME's ADI-648. Also the ADI-648 can
be MIDI remote controlled without any additional line or cabling between computer (MADI card)
and ADI-648.
8.6 Internal Jumper
The internal blue jumper X4 is neither input nor output. It allows to change the voltage level at
the coaxial MADI output. In the lower position, the card generates 600 mVpp (according to the
specification). in case of a very long or 'lossy' coaxial cable, a higher output level might help to
maintain an error-free operation. In the upper position, the output level is 1.2 Vpp.
The HDSP system can play back audio data in supported formats only (sample rate, bit
resolution). Otherwise an error message appears (for example at 22 kHz and 8 bit).
In the audio application being used, HDSP must be selected as output device. This can often
be found in the Options, Preferences or Settings menus under Playback Device, Audio Devices, Audio etc. We recommend using 24-bit resolution for playback, to make full use of the HDSP’s
potential.
We strongly recommend switching all system sounds off (via >Control Panel /Sounds<). Also
HDSP 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 buying a cheap Blaster clone and select this as Preferred Device in >Control
Panel /Multimedia /Audio<.
The RME Driver CD includes step by step instructions for configuring many popular audio
applications, found in the directory \rmeaudio.web\english\techinfo\conf.
The screenshot to the
right shows a typical
configuration dialog as
displayed by a (stereo)
wave editor. After
selecting one of the 32
playback devices,
audio data is sent to
the according audio
channels.
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’. Even at higher buffer
settings in a mixed Audio/MIDI environment, sync problems will not arise because the
Hammerfall DSP always reports the current play position correctly (even while recording essential for chase lock synchronization).
9.2 DVD-Playback (AC-3 / DTS / Multichannel) under MME
AC-3 / DTS
When using popular DVD software player like WinDVD and PowerDVD, their audio data stream
can be sent to any AC-3/DTS capable receiver using the HDSP MADI. For this to work, in most
cases a device of the HDSP MADI has to be selected in 'Control Panel/Sounds and
Multimedia/Audio'. Also check 'use preferred device only'.
You will notice that the DVD software's audio properties now allow to use 'SPDIF Out' or to
'activate SPDIF output'. When selecting these, the software will transfer the non-decoded
digital multichannel data stream to the Hammerfall DSP. Naturally a successful decoding also
requires a MADI to AES converter, converting the playback channels to AES or SPDIF.
Note: AC-3 sounds like chopped noise at highest level.
Multichannel
PowerDVD and WinDVD can also operate as software decoder, sending a DVD's multichannel
data stream directly to any channels of the HDSP MADI. Supported are all modes, from 2 to 8
channels, at 16 bit resolution and 48 kHz sample rate.
For this to work an output wave device of the HDSP has to be selected in 'Control
Panel/Sounds and Multimedia/Audio'. Also check 'use preferred device only'. PowerDVD's
audio properties now lists several multichannel modes. If one of these is selected, PowerDVD
plays back the decoded analog multichannel data via the HDSP MADI.
The device selected as Preferred Playback Device defines the first playback channel. Choosing
ADAT 3/4 and 6-channel mode, playback will happen on channels 3 to 8. Choosing MADI 3/4
and 6-channel mode, playback will happen on channels 3 to 8.
The channel assignment using PowerDVD is:
1 (first chosen playback channel) - Left
2 - Right
3 - Center
4 - LFE (Low Frequency Effects)
5 - SR (Surround Right)
6 - SL (Surround Left)
Note 1: Setting the card to be used as system playback device is against common sense, as
professional cards are not specialized to play back system sounds, and shouldn't be disturbed
by system events. To prevent this, be sure to re-assign this setting after usage, or to disable
any system sounds (tab Sounds, scheme 'No audio').
Note 2: The DVD player will be synced backwards from the HDSP card. This means when
using AutoSync and/or word clock, the playback speed and pitch follows the incoming clock
and sample rate.
Note 3: PowerDVD 5 no longer supports a starting channel other than channel 1.
Note 4: In WinDVD 5, a channel-seperated 5.1 playback using the HDSP requires a change in
the registry. Start regedit, go to HKEY_CURRENT_USER / Software / InterVideo / DVD5 / AUDIOCHAN, and set its value to 4 (hexadecimal). Additionally select 'Waveout' as Audio
Renderer in the Audio configuration dialog. Warning: Changes to the registry are done at your own risk. Danger of complete data loss!
9.3 Low Latency under MME (Buffer Size Adjustment)
Using Windows 95 or 98 the MME buffer size was nothing to worry about. Latencies below 46
ms were not possible. Meanwhile both computers and operating system have become much
more powerful, and with Windows 2000/XP latencies far lower can be used. SAWStudio and
Sonar allowed to use such low settings from the start. Sequoia was updated in version 5.91,
WaveLab in version 3.04.
In the HDSP's Settings dialog the MME buffersize (in fact the DMA buffer size) is set with the
same buttons as the ASIO buffer size. Our test computers allow to use settings down to 64
samples without clicks. Please note that this setting only defines the buffer size of the
hardware. The true and effective latency is configured within the MME application!
Attention: the DMA buffers must not be larger than the application's buffers. This case can
happen unnoticed when using ASIO and MME at the same time (multi-client) and setting
ASIO to 186 ms, while the buffers in the MME application are still set for a lower latency.
Playback will be stuttering and audio will be distorted.
Example: when you set the Hammerfall to 512 you can't use 128 in any program. But setting
DMA to 128 allows to use 128 and all higher values within the software.
Please also note that this is a 'you're welcome to try' feature. We can't guarantee that you will
be able to use 3 or 6 ms with MME. Simply check out by yourself which lowest setting your
system and software allows. Some motherboards with insufficient PCI bandwidth (especially
VIA based) suffer from crackling at settings below 512. Be sure to set the buffer size to 512 or
higher in such a case (or trash the motherboard…).
9.4 Multi-Client Operation
RME audio cards support multi-client operation. This means several programs can be used at
the same time. Also all formats, like ASIO, MME and GSIF can be used simultaneously. The
use of multi-client operation requires to follow two simple rules:
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 channels at the same time.
If for example Cubase uses channels 1/2 (default in Cubase, Master bus), this playback pair
can't be used in Gigasampler/Studio (GSIF) nor under MME (WaveLab etc) anymore (the
inputs can be used at the same time). This is no limitation at all, because TotalMix allows any
output routing, and with this a playback of multiple software on the same hardware outputs.
ASIO Multi-client
RME audio cards 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. The inputs can be used simultaneously.
An exception is our sophisticated tool DIGICheck. It operates like an ASIO host, using a special
technique to access playback channels already occupied. Therefore DIGICheck is able to
perform an analyzis and display of 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 I/O cards always need a valid input signal to start recording.
To take this into account, RME has included two unique features in the HDSP MADI: a
comprehensive I/O signal status display (showing sample frequency, lock and sync status) in
the Settings dialog, and the protective Safe Mode / Frequency function.
If a 48 kHz signal is fed to the input and the application is set to 44.1 kHz, recording is
prohibited. This prevents faulty takes, which often go unnoticed until later on in the production.
Such tracks appear to have the wrong playback rate - the audio quality as such is not affected.
The sample frequency shown in the Settings dialog (see chapter 10, 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’.
With this configuring any suitable audio application for digital recording is simple. After
selecting the required input, Hammerfall DSP displays the current sample frequency. This
parameter can then be changed in the application’s audio attributes (or similar) dialog.
The screenshot to the right shows a typical dialog
used for changing basic parameters such as
sample frequency and resolution in an audio
application.
Any bit resolution can be selected, providing it is
supported by both the audio hardware and the
software. Even if the input signal is 24 bit, the
application can still be set to record at 16-bit
resolution. The lower 8 bits (and therefore any
signals about 96dB below maximum level) are
lost entirely. On the other hand, there is nothing
to gain from recording a 16-bit signal at 24-bit
resolution - this would only waste precious space
on the hard disk.
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 15).
TotalMix also includes a useful automatic real-time monitor function, see chapter 15.8 for
details. Activating record in the application causes the input signal to be routed according to the
current mixer settings.
Currently two solutions exist which enable an automated control of real-time monitoring. ZLM
(Zero Latency Monitoring) allows monitoring in Punch I/O mode - with this the system behaves
like a tape machine. This method has been implemented in all versions of Samplitude (by
SEK’D), and can be activated using the global track option 'Hardware monitoring during Punch'.
The other solution is Steinberg’s ASIO protocol with our ASIO 2.0 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 Record is started.
Configuring the HDSP system is done using its own settings dialog. The panel 'Settings' can be
opened in two different ways:
• by clicking on the hammer icon in the Taskbar's system tray
The mixer of the Hammerfall DSP System (TotalMix) can be opened in two different ways:
• by clicking on the mixer icon in the Taskbar's system tray
The Hammerfall DSP’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:
• Input selection
• Output mode
• Synchronization behaviour
• Input and output status display
• Time code display*
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 the user precise
information about the current status of
the system, and the status of all
signals. ‘SyncCheck’ indicates whether
there is a valid signal for each input
(‘Lock’ or ‘No Lock’), or if there is a valid and synchronous signal (‘Sync’). The ‘AutoSync Ref’
display shows the input and frequency of the current sync source.
*'Time Code' displays time information received from the optional Sync Module.
Quick Boot
All the card's settings described below are stored in a hardware memory, and are loaded
immediately after a power-on of the computer. In clock mode master even the last used sample
rate is set. Directly after switching on the computer, a stable and predictable clock state is
found at the HDSP MADI's outputs. This advanced technology completely eliminates disturbing
noises and clock network problems during power-up or re-boot times.
Frequency verifies the current input signal against the settings in the record program. When deactivated a recording will always be allowed, even with non-valid input signals.
SyncAlign ensures fully sample-aligned channels within MME multitrack software. This option
should only be switched off in case the used software does not work at all with SyncAlign
active.
Input activates redundancy operation. If the current input signal fails, the other input will be
used immediately, provided a valid signal is found there.
TMS activates the transmission of Channel Status data and Track Marker information from the
MADI input signal.
Buffer Size
The setting Buffer Size determines the latency between incoming and outgoing ASIO and GSIF
data, as well as affecting system stability (see chapter 13). Under Windows MME this setting
determines the DMA buffer size (see chapter 9.3).
SyncCheck
‘SyncCheck’ indicates for coaxial and
optical MADI input whether there is a
valid signal (‘Lock’ or ‘No Lock’), or a
valid and synchronous signal (‘Sync’).
The ‘AutoSync Ref’ display shows the
input and frequency of the current sync
source.
MADI In
Defines the input for MADI signal.
'Coaxial' relates to the BNC socket,
'Optical' to the optical input.
MADI Out
Defines the format of the MADI output
signal. MADI can be a 56 or 64 channel
signal. Sample rates higher than 48 kHz
can be transmitted using the normal
48K Frame, or using a native 96K
Frame at the card's output.
Clock Mode
The card can be configured to use its
internal clock (Master), or the clock
source pre-defined via Pref. Sync Ref
(AutoSync).
Pref. Sync Ref
Used to pre-select the desired clock source. If the selected source isn't available the card will
change to the other one. The currently used clock source and sample rate is displayed in the
AutoSyncRef display. The automatic clock selection checks and changes between the clock
sources MADI and word clock.
System Clock
Shows the current clock state of the HDSP system. The system is either Master (using its own
clock) or Slave (AutoSync Ref).
The HDSP MADI has been equipped with AutoSync, an automatic clock source selection,
which adopts the first available input with a valid digital signal as the clock reference input. The
input currently used as sync reference is shown in the AutoSync Ref status field, together with
its sample frequency.
AutoSync guarantees that normal record and record-while-play will always work correctly. In
certain cases however, AutoSync may cause feedback in the digital carrier, so synchronization
breaks down. To remedy this, switch the HDSP’s clock mode over to 'Master'.
Via Pref. Sync Ref (preferred
synchronization reference) a preferred
input can be defined. As long as the
card sees a valid signal there, this
input will be designated as the sync
source, otherwise the other inputs will
be scanned in turn. If none of the
inputs are receiving a valid signal, the
card automatically switches clock
mode to ‘Master’.
Thanks to its AutoSync technique and
lightning fast PLLs, the HDSP is not
only capable of handling standard
frequencies, but also any sample rate
between 25 and 200 kHz. Even the
word clock input, most often used in
varispeed operation, allows any
frequency between 25 kHz and 200
kHz.
The HDSP MADI's outstanding clock
control allows for a synchronization of
the output signal to the input signal not
only at identical sample rates, but also
at half, quarter, double and quad
sample rates. A playback of 192 kHz can easily be synchronized via a 48 kHz clock source.
SyncCheck
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. This is why
digital systems always need a single device defined as ‘master’, which sends the same clock
signal to all the other (‘slave’) devices. RME’s exclusive SyncCheck technology (first
implemented in the Hammerfall) enables an easy to use check and display of the current clock
status. The ‘SyncCheck’ field indicates whether no signal (‘No Lock’), a valid signal (‘Lock’) or a
valid and synchronous signal (‘Sync’) is present at each of the digital clock source inputs. The
‘AutoSync Ref’ display shows the current sync source and the measured frequency.
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.
In the analog domain one can connect any device to another device, a synchronization is not
necessary. Digital audio is different. It uses a clock, the sample frequency. The signal can only
be processed and transmitted when all participating devices share the same clock. If not, the
signal will suffer from wrong samples, distortion, crackle sounds and drop outs.
AES/EBU, SPDIF, ADAT and MADI are self-clocking, an additional word clock connection in
principle isn't necessary. But when using more than one device simultaneously problems are
likely to happen. For example any self-clocking will not work in a loop cabling, when there is no
'master' (main clock) inside the loop. Additionally the clock of all participating devices has to be
synchronous. This is often impossible with devices limited to playback, for example CD players,
as these have no SPDIF input, thus can't use the self clocking technique as clock reference.
In a digital studio synchronisation is maintained by connecting all devices to a central sync
source. For example the mixing desk works as master and sends a reference signal, the word
clock, to all other devices. Of course this will only work as long as all other devices are
equipped with a word clock or sync input, thus being able to work as slave (some professional
CD players indeed have a word clock input). Then all devices get the same clock and will work
in every possible combination with each other.
Remember that a digital system can only have one master! If the HDSP MADI’s clock mode
is set to 'Master', all other devices must be set to ‘Slave’.
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 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. No wonder Superclock did not become a commonly accepted standard.
The actual end of these problems is offered by the SteadyClock technology of the HDSP
MADI. Combining the advantages of modern and fastest digital technology with analog filter
techniques, re-gaining a low jitter clock signal of 11 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.
This is especially true when extracting the word clock out of a MADI signal. Caused by the
MADI format itself, such a signal will have around 80 (!) ns of jitter, which is reduced to about 1
(!) ns by SteadyClock.
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.
To avoid voltage loss and reflections, both the cable itself and the terminating resistor 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.
Ideally all outputs of word clock delivering devices are designed with very low impedance, but
all word clock inputs with high impedance, in order to not weaken the signal on the chain. But
there are also negative examples, when the 75 Ohms are built into the device and cannot be
switched off. In this case the network load is often 2 x 75 Ohms, and the user is forced to buy a
special word clock distributor. Note that such a device is generally recommended for larger
studios. Also, 75 Ohm cable is almost impossible to find these days. 50 Ohm cable is standard
- this will also work as long as the termination resistors are 75 Ohm.
The HDSP MADI’s word clock input can be high-impedance or terminated internally, ensuring
maximum flexibility. If termination is necessary (e.g. because HDSP MADI is the last device in
the chain), activate the switch TERM between the BNC jacks on the Expansion Board so that
the yellow TERM LED lights up.
In case the HDSP MADI resides within a chain of devices receiving word clock, plug a Tadapter into the BNC input jack, and the cable supplying the word clock signal to one end of the
adapter, but 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 terminator plug as
described in the previous paragraph. Some devices (like the HDSP MADI) have switchable 75
Ohm resistors, which saves both T-adapter and terminator.
Due to the outstanding SteadyClock technology of the HDSP MADI, we recommend not to
pass the input signal via T-adapter, but to use the HDSP MADI'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 held at the last valid frequency.
11.3 General Operation
The green ‘Lock’ LED at the Expansion Board will light up as soon as a word clock signal is
detected. Selecting ‘Word Clock’ in the ‘Clock Mode’ field will switch clock control over to the
word clock signal. As soon as there is a valid signal at the BNC jack, 'AutoSync Ref' will display
'Word'. 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.
12. Using more than one Hammerfall DSP
The current drivers support up to three Hammerfall DSP MADI. All cards must be in sync, i.e.
have to receive valid sync information (either via word clock or by using AutoSync and feeding
synchronized signals).
We will use Steinberg’s Cubase VST as an example throughout this chapter. All information
provided can easily be adaptated to other programs.
Start the ASIO software
and select ‘System’ from
the Audio menu. Select
'ASIO Hammerfall DSP
MADI' as the audio I/O
device. The 'ASIO
system control' button
opens the HDSP’s
Settings dialog (see
chapter 10,
Configuration).
Hammerfall DSP
supports 'ASIO Direct
Monitoring' (ADM).
When the sample
frequency is set to 88.2
or 96 kHz (Double Speed
Mode), the number of
available channels is
halved.
13.2 Known Problems
In case the used computer has no sufficient CPU-power and/or sufficient PCI-bus transfer
rates, then drop outs, crackling and noise will appear. We also recommend to deactivate all
PlugIns to verify that these are not the reason for such effects.
Unfortunately some newer UltraATA66 and UltraATA100 hard disk controller (also Raid
controller) seem to violate against the PCI specs. To achieve the highest throughput they hog
the PCI bus, even in their default setting. Thus when working with low latencies heavy drop
outs (clicks) are heard. Try to solve this problem by changing the default setting of the
controller (for example by reducing the 'PCI Bus Utilization').
Another common source of trouble is incorrect synchronization. ASIO does not support
asynchronous operation, which means that the input and output signals must not only have the
same sample frequency, but they must also be in sync. All devices connected to the
Hammerfall DSP 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 GSIF interface of the HDSP MADI allows direct operation with Gigastudio, with up to 32*
channels, 96kHz and 24bit.
Gigastudio requires a lot of the computer’s calculation power. An optimum performance is
achieved with a stand-alone GSIF PC. However, the GSIF latency is always the same as the
one selected for ASIO operation. This can cause performance problems on slower machines
when using GSIF and ASIO at the same time.
Please note that the W2k/XP driver fully supports multi-client operation, including the
combination MME/ASIO. So for example Cubase, Gigastudio and Sonar can be used
simultaneously, provided each of these programs uses its own audio channels exclusively. For
example ASIO could use channels 1/2 and Gigastudio (with GSIF) channels 3/4
simultaneously, and so on.
Simultaneous operation of different programs requires to use different channels. For
example, if Cubase uses tracks 1/2 these tracks can not be used by Gigastudio.
Common Problems
Please note that Gigastudio is running unexpectedly in the background (thus blocking its
assigned audio channels), as soon as the Gigastudio MIDI ports are used – even when
Gigastudio itself hasn't been started. This causes a lot of confusion, as the driver seems to
behave completely buggy, and the user does not recognize the simple reason for it – for
example simultaneous operation of ASIO and GSIF on the same channels.
In case Gigastudio loads well, load gig files too, but won't play at all even when using the virtual
keyboard: Go to Hardware/Routing and select a valid MIDI input port. Note that blank is not
valid, but <none> is.
*The limitation of 32 channels is caused by Gigastudio 2.54. According to Tascam, Gigastudio 3 will support 64 channels.
The Hammerfall DSP system includes a powerful digital real-time mixer. RME’s unique
TotalMix technology allows for nearly unlimited mixing and routing with all inputs and playback
channels simultaneously.
Here are some typical applications for TotalMix:
• setting up delay-free submixes (headphone mixes)
• unlimited routing of inputs and outputs (free utilisation, patchbay function)
• distributing signals to several outputs at a time
• simultaneous playback of different programs over only one stereo channel
• mixing of the input signal to the playback signal (complete ASIO Direct Monitoring)
• integration of external devices (effects etc). in real-time
The block diagram of the TotalMix mixer of the HDSP MADI shows that the record signal
always stays un-altered, but can be passed on as often as desired, even with different levels.
The level meter of inputs and playback channels are connected pre-fader (due to the enormous
routing capabilities). The level meters of the hardware’s outputs are connected post-fader.
The visible design of the TotalMix mixer is mainly determined by the architecture of the HDSP
system:
• Upper row: hardware inputs. The level shown is that of the input signal, i. e. Fader
independent. Per fader and routing window, any input channel can be routed and mixed to
any hardware output (third row).
• Middle row: playback channels (playback tracks of the software). Per fader and routing
window, any playback channel can be routed and mixed to any hardware output (third row).
• Lower row: hardware outputs. Because they refer to the output of a subgroup, the level can
only be attenuated here (in order to avoid overloads), routing is not possible. This row has
two additional channels, the analog outputs.
Every single channel has various elements:
Input and playback channels each have a mute and solo button.
Below each there is the panpot, realized as indicator bar (L/R) in order to save space.
In the window below this, the present level is displayed in RMS or Peak, being
updated about every half a second. Overs are indicated here by an additional red dot.
Then comes the fader with a levelmeter. The meter shows both peak values (zero
attack, 1 sample is enough for displaying full scale) 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, the black area shows the current routing
target. Selecting one or more channels is done by clicking on the white label which
turns yellow then.
In the following chapters we will explain all functions of the mixer surface step by step. 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. The faders in the upper row are set to maximum attenuation (called
m.a. in the following), so there is no monitoring of the input channels.
We will now create a submix on outputs 63/64 (headphones). Please start
a multitrack playback and connect a headphone to the card. In playback
channel 1 (labeled 'Out 1'), click onto the routing window below the label. A
list pops up, showing a checkmark in front of '1+2'. Click onto '63/64'. The
list disappears, the routing window no longer shows '1+2', but '63/64'. 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 channels 63 and 64, you can
also see the level of what you are hearing. 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 63 and 64. 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
outputs 63/64 as well.
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. Press the
Ctrl-key and click into the routing window of 'Out 3' with the key pressed.
The routing list pops up with a checkmark at '3+4'. Click onto '63/64'. Now,
channel 4 has already been set to '63/64' 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 window. You'll now see two checkmarks, one at
'3+4', the other one at '63/64'. Click onto '61/62'. The window disappears,
fader and panpot jump to their initial values, the signal can now be routed
to these channels. You can continue, 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 mix has not changed, while you
were routing the channel to other outputs and setting different gain values.
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.
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 '63/64' in
the routing window, pull the fader down, open the routing window again the checkmark is gone.
Such a wide range of possibilities make it difficult to maintain the overview. Because practically
all hardware outputs can be used for different submixes, as shown (up to 32 completely
independent stereo submixes, 16 4-channel submixes etc.). And when opening the routing
windows you might see an army of checkmarks, but you don't get an overwiev, i.e., how the
signals come together and where. This problem is removed by the view mode 'Submix'. In this
mode, all routing windows 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
'63/64').
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.
15.4 Mute and Solo
Mute works pre-fader, thus mutes all active routings of the channel. As soon as any Mute
button is pressed, the Master Mute button lights up in the quick access area. It can switch all
selected mutes off and on again. You can comfortably make mute groups to activate and
deactivate this way.
The same holds true for the Solo and the Master Solo buttons. Solo is working as a solo-inplace. As soon as one Solo button is pressed, all other Mute buttons are activated and light up.
15.5 Hotkeys
TotalMix knows only a few, but very effective key combinations, that make setting the mixer up
considerably easier and faster. The Shift-key for the fine-mode for faders and panpots has
already been mentioned. But 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, -6
dB for the hardware outputs.
• 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
corresponding factory default preset
• Clicking the Card 2 button while holding down Ctrl opens a second TotalMix window.
The faders can also be moved pairwise, corresponding to the stereo-routing settings. This can
be achieved by pressing the Alt-key and is especially comfortable when setting the SPDIF and
analogue output level. Even the Panoramas can be operated with Alt, from stereo through
mono to inversed channels. But 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 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 you want to set several faders to m.a. for instance, 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 each of Input, Playback and Output channels
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 and the Meter
Bridge).
Hotkey M toggles Master Mute on/off (and with this performs a global mute on/off). Hotkey X
toggles the Matrix view on/off (see chapter 16), hotkey T the mixer view.
Further hotkeys are available to control the configuration of the Level Meter (see chapter
15.10):
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
15.6 The Quick Access Panel
This section includes additional options, further improving the handling of TotalMix. The Master
button for Mute and Solo has already been described, they allow for group-based working with
these functions.
In the View section the single rows can be made visible or invisible. If the inputs are not
needed for a pristine playback mix, the whole upper row falls out of the picture after a click on
the input button. If the hardware outputs don't interest you either, the surface can thus be
reduced to the playback channels to save space. All combinations are possible.
Submix sets all routing windows to the same selection as described before. Deactivating
Submix automatically recalls the previous view.
The mixer can also 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 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. Just try it: 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 the present 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.
Also here, RME's love for details can be seen. 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 was 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 and
'Open file'. Mixer settings can of course be saved the usual way, and with long
file names.
Up to three Hammerfall DSP MADI can be used simultaneously. The Card buttons switch
between the cards. Holding down Ctrl while clicking on button Card2 will open a second
window, instead of replacing the current window content.
The number of channels is reduced automatically when entering Double or Quad Speed mode
(96 / 192 kHz). The display is adjusted accordingly, and all fader settings remain stored (even
the invisible ones).
TotalMix includes 8 factory presets, stored within the program. But the presets can be changed
at any time, because TotalMix stores and reads the changed presets form the files
preset11.mix to preset81.mix. These files are found in the hidden directory Documents and
Settings, <Username>, Local Settings, Application Data, RME TotalMix. The first number
indicates the current preset, the second number the current card/system.
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 anytime.
The original factory presets can be reloaded by holding down the Ctrl-key and clicking on any
preset button.
Using 64, 88.2 and 96 kHz, TotalMix loads special factory presets with a submix on channels
31/32, for a working monitoring in these modes.
The 8 factory presets offer not only a useful functionality for TotalMix, but also a pretty good
base to modify them to your personal needs.
Preset1.mix
All channels routed 1:1.
Details: All inputs maximum attenuation (m.a.). All playback channels 0 dB, routet to the same
output. All outputs 0 dB. All channels prepared for all routings to left/right panning. Level
display RMS +3 dB not activated.
Note: This preset is Default, offering the standard functionality of a I/O-card.
Preset2.mix
Details: All channels routed 1:1, playback monitoring plus input monitoring 1:1.
Preset3.mix
Details: All channels routed 1:1. No playback and no input monitoring. All faders down.
Preset4.mix
Details: All channels routed 1:1. As preset 2, but all inputs muted.
Note: This preset is default for ZLM and MME Mix/Replace monitoring. The factory preset 4 will
also be loaded by a click on Load Def.
Preset5.mix
Details: All channels routed 1:1, playback monitoring. Submix of all playback channels to
channel 63/64 (headphone monitoring). Hardware output 63/64 selected and faders at –12 dB.
Preset6.mix
Details: as preset 5, but submix of all input channels to channels 63/64 (headphone monitoring.
Preset7.mix
Details: as preset 5, but submix of all input and playback channels to channels 63/64
(headphone monitoring). View Submix active.
Preset8.mix
Description: Panic. As Preset 4, but also all playback channels muted (no output signal)
The Monitor section of the Quick Access Panel is only valid for our Windows MME driver, i.e.
when using programs like WaveLab, Soundforge, Sonar or Samplitude.
Monitor offers two advanced automated monitoring solutions. Monitoring will be controlled
either by special commands directly from the recording software
(Samplitude/Sequoia/SAWStudio, mode ZLM), or by the recording state itself (mode
Mix/Replace).
The basic method used is as simple as it is clever: ZLM and record are controlling the Mute
buttons of the corresponding channels. For this to work, Mute must be activated on the
record's Input channel. The fader must not be set to m.a..
A click on Load Def. will load a template (factory preset
4), which can be used to verify and test this functionality.
But it is also possible to use any other mixer state, as
long as the recording channels are muted manually.
As soon as a recording starts, the corresponding
channels will be unmuted, i.e. the input signal will be
processed according to the current mixer settings.
TotalMix lets you check the whole process visually, as
the Mute buttons in TotalMix will be switched on and off
automatically.
In Mix mode the input signal will be mixed on the outputs
when record is active. In Replace mode the Mute button
of the corresponding playback channel will be activated,
so that the input signal replaces the playback signal.
ZLM is a special function for tape machine style
monitoring when doing punch-ins and outs. For this to
work the option 'Hardware monitoring during punch' has
to be activated in Samplitude/Sequoia. Then at each
punch-in the corres-ponding Mute buttons will be
deactivated, at punch-out they will be reactivated.
All settings can be changed and configured in real-time.
15.9 Menu Options
Always on Top: When active (checked) the window of TotalMix will always be on top of the
Windows desktop.
Note: This function may result in problems with windows containing help text, as the TotalMix
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 prest will not be
used. The routing will be activated, but the window will not change
ASIO Direct Monitoring: When de-activated any ADM commands will be ignored by TotalMix.
In other words, ASIO Direct Monitoring is globally de-activated.
Having set a new standard with the level meters of DIGICheck, Hammerfall DSP goes even
further: The calculation of the Peak, RMS and Over is realized in hardware, in order to be
capable of using them independent of the software in use, and to significantly reduce the CPU
load.
The level meters integrated in TotalMix - considering their size - cannot be compared with the
HDSP Meter Bridge (chapter 21.2). Nevertheless they already include many useful functions.
Peak and RMS is displayed for every channel. 'Level Meter Setup' (Menu Options or F2) or
direct keyboard entry (hotkeys) makes various options available:
• 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. RMS shows 3 dB less for
sine signals. This is mathematically correct, but not
very reasonable for a level meter. Therefore, we
had corrected DIGICheck's RMS display by 3 dB, a
full scale sine signal shows both 0 dBFS Peak and
RMS. This setting also yields directly readable
signal-to-noise values, while other applications
(like WaveLab) will show a value 3 dB better than
actual (because the reference is not 0 dB, but -3
dB).
The value displayed in the text field is independent
of the setting 40/60 dB, it represents the full 24 bit
range of the RMS measurement, thus making
possible a SNR measurement 'RMS unweighted',
which you would otherwise need extremely
expensive measurement devices for. An ADI-8 DS
connected to an ADI-648 and then to the MADI
card will therefore show around -113 dBFS on all 8
channels.
This level display will constantly bring the reduced dynamic range of your equipment, maybe of
the whole studio, in front of your eyes. Nice to have everything 24 bit - but still noise and hum
everywhere in the range around -90 dB or worse... sorry, but this is hard reality. The up-side
about it is that TotalMix allows for constantly monitoring the signal quality without effort. Thus it
can be a valuable tool for sound optimization and error removal in the studio.
Measuring SNR (Signal to Noise) requires to press R (for RMS) and 0 (for referring to 0
dBFS, a full scale signal). The text display will then show the same value as an expensive
measurement system, when measuring ‘RMS unweighted’.
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
mixer looks a 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.
16.1 Elements of the Surface
The visible design of the TotalMix Matrix is mainly determined by the architecture of the HDSP
system:
• 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.
16.2 Operation
Using the Matrix is a breeze. It is very easy to indentify the current crosspoint, because the
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 In1 / 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
sources, and the upper side, representing the hardware outputs. If you move a fader in row 1 or
2 in TotalMix, only the specific levels (max. 2) of this routing will change in the Matrix. But
moving a fader in row 3 will make all vertically activated levels move at once (for example
63/64, analog output).
A gain field marked orange indicates activated Mute status. Mute can only be changed in the
mixer view.
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 on 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 software outputs on the analog output (equals factory preset
5). Setting up such a submix is fast and easy. But how to check at a later time, that all settings
are still exactly the way you wanted them to be?
So far the only way to check that TotalMix is correctly set up this way, is to activate Submix
view, step through all existing software outputs, and have 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
Ctrl-mouse, and move on. You can see in the desk view how level and pan changes
automatically when performing the second (fourth...) setting.
17. TotalMix Super-Features
17.1 ASIO Direct Monitoring
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 calculation, 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: if you move the mouse so that any channel reaches upper or lower maximum position,
and release the mouse button, the relative position is lost.
Tip: gang some submixes and watch all routing levels change like crazy in the Matrix view.
17.3 Copy Routings to other Channels
TotalMix allows to copy complete routing schemes of inputs and outputs.
Example 1: You have input 5 (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 5, open the menu Edit. It shows 'Copy In 5'. Now select the
desired new input, for example In 8. The menu now shows 'Paste In 5 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 view open 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.
17.4 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!
MADI, the serial Multichannel Audio Digital Interface, has been defined already in 1989 as an
extension of the existing AES3 standard following several manufacturers' wish. The format also
known as AES/EBU, a balanced bi-phase signal, is limited to two channels. Simply put, MADI
contains 28 of those AES/EBU signals in serial, i. e. after one another, and the sample rate can
still even vary by +/-12.5%. The limit which cannot be exceeded is a data rate of 100Mbit/s.
Because an exact sampling frequency is used in most cases, the 64 channel mode was
introduced officially in 2001. It allows for a maximum sample rate of 48 kHz + ca. 1%,
corresponding to 32 channels at 96 kHz, without exceeding the maximum data rate of 100
Mbit/s. The effective data rate of the port is 125 Mbit/s due to additional coding.
Older devices understand and generate only the 56 channel format. Newer devices often work
in the 64 channel format, but offer still no more than 56 audio channels. The rest is being eaten
up by control commands for mixer settings etc.. The ADI-648 and the HDSP MADI show that
this can be done in a much better way, with an invisible transmission of 16 MIDI channels and
the MADI signal still being 100% compatible.
For the transmission of the MADI signal, proved methods known from network technology were
applied. Most people know unbalanced (coaxial) cables with 75 Ohms BNC plugs, they are not
expensive and easy to get. The optical interface is much more interesting due to its complete
galvanic separation, but for many users it is a mystery, because very few have ever dealt with
huge cabinets full of professional network technology. Therefore here are some explanations
regarding 'MADI optical'.
• The cables used are standard in computer network technology. They are thus not at all
expensive, but unfortunately not available in every computer store.
• The cables have an internal fibre of only 50 or 62.5 µm diameter and a coating of 125 µm.
They are called network cables 62.5/125 or 50/125, the former mostly being blue and the
latter mostly being orange. In most cases, they are not (!) glass fibre cables, but plastic fibre
cables (POF, plastic optical fibre).
• The plugs used are also an industry standard and called SC. Please don't mix them up with
ST connectors, which look similar to BNC connectors and are being screwed. Plugs used in
the past (MIC/R) were unnecessarily big and are not being used any longer.
• The cables are available as a duplex variant (2 cables being glued together) or as a
simplex variant (1 cable). The HDSP MADI's opto module supports both variants.
• The transmission uses the multimode technique which supports cable lengths of up to
almost 2 km. Single mode allows for much longer distances, but it uses a completely
different fibre (8 µm). By the way, due to the wave-length of the light being used (1300 nm),
the optical signal is invisible to the human eye.
The Hammerfall DSP MADI's SteadyClock technology guarantees an excellent performance in
all clock modes. Its highly efficient jitter suppression refreshes and cleans up any clock signal,
and provides it as reference clock at the word clock output.
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, caused
by the time resolution of 125 MHz within
the format. But common jitter values for
other devices are 5 ns, while a very
good clock will have less than 2 ns.
The picture to the right shows the MADI
input signal with 80 ns of jitter (top
graph, yellow). Thanks to SteadyClock
this signal turns into a clock with less
than 2 ns jitter (lower graph, blue). The
cleaned and jitter-freed signal can be
used as reference clock for any
application, without any problem. The signal processed by SteadyClock is of course not only
used internally, but also available at the card's word clock output.
20. Hotline - Troubleshooting
20.1 General
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.
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 Hammerfall DSP 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
• 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’.
• 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).
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. This is
no issue when using dual CPU systems.
20.2 Installation
More information on installation problems (which fortunately are very seldom, thanks to Plug
and Play), can be found in the Tech Info 'Installation problems'. It is located in the directory
\rmeaudio.web\techinfo on the RME Driver CD.
HDSP MADI is normally found in the Device Manager (>Settings/Control Panel/System<),
category 'Sound-, Video- and Gamecontroller'. A double click on ' Hammerfall DSP MADI'
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.rme-
audio.com, section FAQ, Hardware Alert: about incompatible hardware.
The dialog 'New hardware component found’ does not appear:
• Check whether 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 is worth checking the ‘Resources’ tab
anyway.
• Check whether the Hammerfall DSP has been selected as current ASIO device.
The DIGICheck software is a unique utility developed for testing, measuring and analysing
digital audio streams. Although the DIGICheck software is fairly self-explanatory, it still includes
a comprehensive online help. DIGICheck 4.0 operates as multi-client ASIO host, and can
therefore be used in parallel to any software, be it MME, ASIO or GSIF, both inputs and even
outputs (!). The following is a short summary of the available functions:
• Level Meter. High precision 24-bit resolution, 2/8/26 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.
• Vector Audio Scope. World wide unique Goniometer showing the typical afterglow of a
oscilloscope-tube. Includes Correlation meter and level meter.
• 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 and Vector Audio Scope as one window function.
• Channel Status Display. Detailled analyzis and display of SPDIF and AES/EBU Channel
Status data.
• Completely multi-client. Open as many measurement windows as you like, on any
channels 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.
21.2 HDSP Meter Bridge 2.0
The Hammerfall DSP Meter Bridge is a unique, highly flexible and handy tool for level
metering. As opposed to DIGICheck, the HDSP Meter Bridge receives all level data directly
from the hardware. Thus the Meter Bridge can not only run in parallel to any other program, but
as Peak, Over and RMS calculations are performed directly in hardware, the CPU load caused
is limited to the graphics routines – and is near zero on todays computers! Available functions
are:
• 2-channel Level Meter. High precision 24-bit resolution. Application examples: Peak level
measurement, RMS level measurement, over-detection, dynamic range and signal-to-noise
ratios, RMS to peak difference (loudness), long term peak measurement, input check.
• 8-channel Level Meter. As 2-channel Level Meter, but combined Peak/RMS display.
• Global Level Meter. As 8-channel Level Meter, but displays all channels of a card.
• Completely multi-client. Open as many measurement windows as you like, on any
channels and inputs or outputs!
Although the HDSP Meter Bridge is fairly self-explanatory, it still includes a comprehensive
online help. To start press F2 and F3, the most important hotkeys.
To install the HDSP Meter Bridge, go to the \HDSP Meter Bridge directory on the RME Driver
CD and run setup.exe. Follow the instructions prompted on the screen.
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:
Synchronization II (DIGI96 series)
Digital audio synchronization - technical background and pitfalls.
Installation problems
Problem descriptions and solutions.
Information on driver updates
Lists all changes in the drivers.
Configuring Logic, Samplitude, Cubase, Cakewalk, Sonar and SAWPlus32
Step by step instructions for use with RME cards.
DIGICheck: Analysis, tests and measurements with the DIGI96 series
A description of DIGICheck, including technical basics.
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
HDSP System: TotalMix - Hardware and Technology
HDSP System: TotalMix - Software, features, operation
The digital mixer of the Hammerfall DSP in theory and practise
Each individual Hammerfall DSP undergoes comprehensive quality control and a complete test
in a PC environment at RME before shipping. This may cause very slight signs of wear (if it
looks like it was used one time before - it was). 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.
RME’s replacement service within this period is handled by the retailer. If you suspect that your
card 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.
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.
25. 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 load a complete copy of the RME website
from the RME Driver CD (in the \rmeaudio.web directory) into your browser.
Trademarks
All trademarks, registered or otherwise, are the property of their respective owners. RME,
DIGI96, SyncAlign, ZLM, SyncCheck, DIGICheck and Hammerfall are registered trademarks of
RME Intelligent Audio Solutions. TotalMix, Intelligent Clock Control and TMS 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 98/SE/ME and Windows
2000/XP 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. Pentium is a registered trademark of Intel Corp.
Copyright Matthias Carstens, 11/2003. Version 1.1
Current driver version: W2k: 2.0
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 EN55022 class B and EN50082-1
norms for digital devices, according to the European Council directive on counterpart laws in
the member states relating to electromagnetic compatibility (EMVG).
FCC
This device has been tested and found to comply with the requirements listed in FCC
Regulations, part 15 for Class ‘B’ digital devices. Compliance with these requirements provides
a reasonable level of assurance that your use of this product in a residential environment will
not result in harmful interference with other electronic devices.
This equipment generates radio frequencies and, if not installed and used according to the
instructions in the User’s Guide may cause interference harmful to the operation of other
electronic devices.
Compliance with FCC regulations does not guarantee that interference will not occur in all
installations. If this product is found to be the source of interference, which can be determined
by turning the unit off and on again, please try to eliminate the problem by using one of the
following measures:
• Relocate either this product or the device that is being affected by the interference
• Use power outlets on different branch circuits, or install AC line filters
• Contact your local retailer or any qualified radio and television engineer
When connecting external devices to this product, compliance to limits for a Class ‘B’ device
requires the use of shielded cables.
FCC compliance statement: Tested to comply with FCC standards for home or office use.