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User’s Guide
ADI-2 Pro
FS R
Conversion done right
2 Channels Analog / Digital Converter
4 Channels Digital / Analog Converter
AES / ADAT / SPDIF Interface
32 Bit / 768 kHz Digital Audio
USB 2.0 Class Compliant
2 Extreme Power Headphone Outputs
Digital Signal Processing
Advanced Feature Set
Extended Remote Control
32 Bit / 768 kHz
Hi-Res Audio
SteadyClock FS
SyncCheck
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User’s Guide ADI-2 Pro FS R - v 3.3
General
1 Introduction ............................................................... 5
2 Package Contents ..................................................... 5
3 System Requirements ............................................... 5
4 Brief Description and Characteristics ..................... 6
5 First Usage - Quick Start
5.1 Connectors and Controls......................................... 7
5.2 Quick Start............................................................... 8
5.3 Operation at the unit ................................................ 8
5.4 Overview Menu Structure ........................................ 9
5.5 Playback ................................................................ 10
5.6 Analog Recording .................................................. 10
5.7 Digital Recording ................................................... 10
6 Power Supply ........................................................... 11
7 Firmware Update ..................................................... 11
8 Features Explained
8.1 Extreme Power Headphones Outputs ................... 12
8.2 Dual Phones Outputs ............................................ 13
8.3 5-band Parametric EQ .......................................... 13
8.4 Bass / Treble ......................................................... 14
8.5 Loudness ............................................................... 14
8.6 SRC (Sample Rate Conversion) ........................... 15
8.7 Crossfeed .............................................................. 15
8.8 DSP Limitations ..................................................... 16
Basic and Stand-Alone Operation Details
9 RME Multi-Remote Control (MRC)
9.1 Keys and Functions .............................................. 18
9.2 Other Remote Controls ........................................ 19
10 Front Panel Controls
10.1 Keys ..................................................................... 19
10.2 Encoders .............................................................. 19
11 VOL ........................................................................... 20
12 I/O
12.1 Analog Input
12.1.1 Settings ......................................................... 21
12.1.2 Parametric EQ .............................................. 22
12.2 Main Output 1/2
12.2.1 Settings ......................................................... 22
12.2.2 Bass/Treble .................................................. 24
12.2.3 Loudness ...................................................... 24
12.3 Phones Output 3/4 ............................................... 25
13 EQ ............................................................................. 25
14 SETUP
14.1 Options
14.1.1 SPDIF / Remap Keys ................................... 27
14.1.2 Clock ............................................................ 28
14.1.3 Device Mode / DSD ...................................... 28
14.1.4 Phones ......................................................... 29
14.1.5 Display .......................................................... 30
14.2 Load/Store all Settings ......................................... 30
15 Top Screens
15.1 Global Level Meter ............................................... 31
15.2 Analyzer ................................................................ 31
15.3 State Overview ..................................................... 32
15.4 Dark Volume ........................................................ 33
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16 Warning Messages.................................................. 34
17 Modes
17.1 Auto ...................................................................... 36
17.2 Preamp ................................................................ 37
17.3 AD/DA Converter ................................................. 38
17.4 USB ...................................................................... 39
17.5 Digital Through ..................................................... 41
17.6 DAC...................................................................... 42
18 Balanced Phones Mode .......................................... 43
19 DSD Operation
19.1 General ................................................................ 44
19.2 Direct DSD ........................................................... 44
19.3 DSD Playback ...................................................... 45
19.4 DSD Record ......................................................... 45
19.5 DSD Level Meter .................................................. 46
19.6 Beyond… ............................................................. 46
Inputs and Outputs
20 Analog Inputs .......................................................... 48
21 Analog Outputs
21.1 General ................................................................ 48
21.2 Line Out TRS 1/2 ................................................. 49
21.3 Line Out XLR 1/2 ................................................. 49
21.4 PH Out 1/2 ........................................................... 49
21.5 PH Out 3/4 ........................................................... 50
22 Digital Connections
22.1 AES ...................................................................... 50
22.2 SPDIF................................................................... 51
22.3 ADAT ................................................................... 52
Installation and Operation - Windows
23 Driver Installation .................................................... 54
24 Configuring the ADI-2 Pro
24.1 Settings Dialog ..................................................... 55
24.2 Clock Modes - Synchronization ............................ 56
25 Operation and Usage
25.1 Playback ............................................................... 56
25.2 DVD Playback (AC-3 / DTS) ................................ 57
25.3 Multi-client Operation ........................................... 57
25.4 Multi-interface Operation ...................................... 57
25.5 ASIO..................................................................... 58
26 DIGICheck Windows ............................................... 58
Installation and Operation – Mac OS X
27 General ..................................................................... 60
27.1 Configuring the ADI-2 Pro .................................... 60
27.2 Clock Modes - Synchronization ............................ 61
27.3 Multi-interface Operation ...................................... 61
28 DIGICheck Mac ........................................................ 61
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User’s Guide ADI-2 Pro FS R - v 3.3
Installation and Operation – iOS
29 General ..................................................................... 64
30 System Requirements ............................................. 64
31 Setup......................................................................... 64
32 Supported Inputs and Outputs .............................. 64
Technical Reference
33 Technical Specifications
33.1 Analog Inputs ....................................................... 66
33.2 Analog Outputs ..................................................... 66
33.3 Digital Inputs ......................................................... 67
33.4 Digital Outputs ...................................................... 68
33.5 Digital ................................................................... 68
33.6 General ................................................................. 68
33.7 Connector Pinouts ................................................ 69
34 Technical Background
34.1 Lock and SyncCheck ........................................... 70
34.2 Latency and Monitoring ........................................ 71
34.3 USB Audio (Windows) .......................................... 72
34.4 M/S-Processing .................................................... 73
34.5 Emphasis ............................................................. 74
34.6 Balanced Phones Mode ....................................... 75
34.7 SteadyClock ......................................................... 77
34.8 ADI-2 Pro as Hardware I/O for Measurements .... 78
34.9 Noise Level in Hi-Speed Modes ........................... 80
34.10 AD Impulse Responses ........................................ 81
34.11 DA Impulse Responses ........................................ 82
34.12 AD/DA Frequency Response ............................... 83
34.13 AD Filter Curves ................................................... 84
34.14 DA Filter Curves 44.1 kHz .................................... 84
34.15 Loudness .............................................................. 85
34.16 Bass / Treble ........................................................ 85
34.17 Distortion Measurements ..................................... 86
34.18 Extreme Power Charts ......................................... 88
34.19 Phones Distortion Comparison ............................ 89
34.20 Impedance based Level Meters PH 1-4 ............... 89
34.21 Digital Volume Control .......................................... 90
34.22 Bit Test ................................................................. 92
34.23 Operation in the Hi-Fi Environment ...................... 93
Miscellaneous
35 Accessories ............................................................. 98
36 Warranty ................................................................... 98
37 Appendix .................................................................. 99
38 Declaration of Conformity .................................... 100
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User’s Guide ADI-2 Pro FS R – v 3.3
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1. Introduction
RME’s ADI-2 Pro is a true milestone in many ways. Looking at the multitude of AD/DA converters, USB DACs and dedicated headphone amps available, RME developers felt they all lacked
obvious features that are unavoidable to enjoy operation as well as when listening to music. And
while many of those devices claim to use the latest state-of–the–art whatsoever converter chip,
serious magazines and RME staff were repeatedly disappointed to find that in the end the stellar
technical data published in ads and datasheets were nowhere to be found.
With the ever growing popularity of headphones and latest AD/DA chips pushing technical data
further, the time was right for a new RME gem. A unit with the industry’s biggest footprint per
feature ratio, with specs that are as real as RME‘s reputation, a feature set that is unheard of,
useful features that for unknown reasons no one else implemented, and two extremely powerful
headphone outputs, that will be your new reference in accuracy and dynamic range.
Here it is – the ADI-2 Pro, the little wonder, a host of devices all put together into one unit, with a
simple and mostly automated way of using it right out of the box:
A high-end AD/DA converter in professional studio quality
A double headphone amplifier in true high-end quality
A USB DAC like no other - the most versatile and capable one around
A high-end AD/DA frontend and headphone amp for iPad and iPhone
An AD/DA frontend for measurement systems at up to 768 kHz sample rate
A multi-format converter (AES, SPDIF, ADAT) with monitoring
An SPDIF/ADAT playback system
A DSD record and playback solution
All there is left to say now is: Enjoy!
2. Package Contents
ADI-2 Pro FS R
Remote control with battery (MRC)
Manual
External switched power supply, lockable connector, DC 12 V 24 W
Power cord
Digital breakout cable AES/SPDIF (BO968)
Quick start guide
3. System Requirements
General:
Power supply 12V DC, 1.5 A or up
For computer based operation:
Windows 7 or up, Intel Mac OS X (10.6 or up)
1 USB 2.0 port or USB 3 port
Computer with at least Intel Core i3 CPU
For iOS based operation:
iPhone or iPad with iOS 7 or up
Dock or Lightning to USB adapter
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4. Brief Description and Characteristics
The ADI-2 Pro is a 2-channel analog input to digital and 4-channel digital to analog output converter in a half-rack (9.5") enclosure of 1 U height. Latest 32 bit / 768 kHz converters offer up to
124 dBA signal to noise ratio. This value is not only printed in the brochure – it is what the unit
achieves in real-world operation.
Reference class tech specs throughout are combined with an unprecedented feature set. A
powerful DSP adds all kinds of useful audio processing, including 5-band parametric EQ, fast
Bass/Treble adjustment, Crossfeed, and a new concept in Loudness sound control.
Operation is quick and easy through 3 encoders with push button function and 4 more buttons to
access dedicated menus. The unit remembers all current settings, even the menu position. Ad-
ditionally the unit’s whole setup as well as equalizer settings can be stored under individual
names.
A high resolution IPS panel for the graphical operation surface eases operation even more, and
displays further functions provided by the DSP, namely Peak level meters, a 30-band analyzer in
DIGICheck biquad filter technology, and a State Overview screen listing the current states of
SPDIF, AES, USB and clock.
The digital inputs SPDIF coaxial (or optical) and AES operate simultaneously. An additional
Sample Rate Converter decouples the SPDIF or AES clock for even simpler setups, and also
supports up- and down-sampling of the input signals. SPDIF optical also supports 2 channels of
ADAT operation, at up to 192 kHz.
When used as USB interface, the Class Compliant UAC 2 mode can be set to Stereo or Multichannel. Multi-channel mode turns the ADI-2 Pro into a 6 channel (Analog 1/2, AES, SPDIF)
record and 8 channel (Analog 1/2/3/4, AES, SPDIF) playback audio interface, that even works
as iPad front-end up at up to 192 kHz sample rate. In stereo mode sample rates up to 768 kHz
are supported, for high resolution recordings or PCM, DXD and DSD record/playback.
The servo balanced analog inputs and dedicated balanced and unbalanced outputs are fitted
with both XLR and 1/4" TRS/TS jacks. The unit uses a fully balanced and DC-coupled circuit
design, for highest phase accuracy at lowest roll-off. The only capacitors in the whole signal
path, non-polar MUSE audio capacitors from Nichicon, reside directly at the input of the unit (DC
protection).
The two Extreme Power headphone outputs provide reference sound and headroom. RME’s
Advanced Balanced mode not only adds balanced phones operation to the feature list, but also
premiers a new concept on improving the balanced mode even further.
To maintain the full dynamic range within the best operating level, discrete 4-stage reference
level settings were realized for maximum dynamic range (+4, +13, +19, +24 dBu). Also available
are 0 to +6 dB digital trim for fine input sensitivity adjustment in steps of 0.5 dB.
The ADI-2 Pro supports sample rates between 44.1 kHz and 768 kHz. Furthermore, RME's
SteadyClock FS guarantees exceptional performance in all clock modes. Thanks to a highly
efficient jitter suppression, the AD- and DA-conversion always operates on highest sonic level,
being completely independent from the quality of the incoming clock signal.
The ADI-2 Pro shines in both studio and home usage. Its click- and noise-free on/off operation
and a comfortable, illuminated standby button add to the soft, modern desktop design.
Mobile and galvanically separated usage is possible through a 12V connector for easy battery
connection.
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User’s Guide ADI-2 Pro FS R – v 3.3
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!
!
5. First Usage – Quick Start
5.1 Connectors and Controls
The front of the ADI-2 Pro has 3 hi-precision rotary encoders with push function, 4 buttons, a
standby power button, a high resolution IPS display, and two TRS headphone outputs.
The output channels 1/2 and 3/4 feed two phones outputs via two independent Extreme Power
driver circuits, optimized for both high and low impedance headphones. Their unbalanced output
signal is of highest quality. With 120 dBA SNR there is no audible hum and noise at those outputs.
In case a phones output is to be used as line output, an adapter TRS plug to RCA phono plugs,
or TRS plug to two TS plugs is required.
The rear of the ADI-2 Pro has 2 servo-balanced analog inputs on combo XLR/TRS sockets, 2
TS sockets as unbalanced outputs, 2 XLR sockets as balanced outputs, TOSLINK optical I/O, a
USB socket, and a lockable power socket. An included breakout cable is connected to the DB-9
socket and provides AES I/O via XLR and SPDIF coaxial I/O via RCA.
The ADI-2 Pro has two analog line inputs that can operate with levels up to +24 dBu. The electronic input stage uses a servo balanced design which handles unbalanced and balanced signals correctly, automatically adjusting the level reference.
When using unbalanced cables with the XLR inputs, pin 3 of the XLR jack should be connected to ground. Otherwise noise may occur, caused by the unconnected negative input of
the balanced input.
The analog inputs 1/2 include 6 dB digital gain control via DSP, and a choice of +4 dBu, +13
dBu, +19 dBu or +24 dBu as reference level, performed in the analog domain.
The short circuit protected, low impedance XLR line outputs do not operate servo balanced! When connecting unbalanced equipment via XLR, make sure pin 3 of the XLR output
is not connected. A connection to ground might cause a decreased THD (higher distortion)
and increased power consumption!
Optical I/O (TOSLINK): The unit automatically detects SPDIF or ADAT input signals. The optical
output can operate as ADAT or SPDIF output, depending on the current setting in the Setup
menu (Options, SPDIF / Remap Keys). Note that only channels 1/2 from the ADAT stream can
be accessed. SMUX and SMUX4 (up to 192 kHz) are still supported.
USB 2.0: Standard USB socket for connection to the computer. The ADI-2 Pro operates as
Class Compliant device, in either 2-channel or 8-channel mode (configured in Setup). It can be
directly used with Mac OS X and iOS (iPad, iPhone). For Windows the RME MADIface series
driver adds WDM and ASIO.
Socket for power connection. This socket supports locking type jacks as found on the included
RME DC power supply. After inserting the connector carefully turn it by 90° so that it locks.
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User’s Guide ADI-2 Pro FS R - v 3.3
5.2 Quick Start
Connect the unit to the included power supply and push the Standby button to start. The ADI-2
Pro ships with Basic Mode Auto activated (SETUP – Options – Device Mode / DSD – Basic
Mode). The input selection for SPDIF (coaxial or optical), and the source selection for Phones
Out 3/4 are also set to Auto, the SRC is activated for the SPDIF input. With Auto active the unit
will turn into different modes depending on connected cables:
Preamp: Analog in to Analog out (internal digital out to in). This mode is active when no digi-
tal input signal and no USB is detected.
AD/DA: Converter Mode, analog in to all digital outs, digital in to all analog outs. This mode
becomes active as soon as a digital signal is attached. This signal will also become the signal
source. In case more than one digital signal is found the user has to manually select the
source to be monitored. The SRC is active as default and assigned to SPDIF. Clock mode is
slave to AES in, but will adapt to the detected source (Auto).
USB: or interface mode (also USB DAC). If USB is detected all inputs are routed to USB, all
outputs are fed from USB. USB has priority over the converter mode. In Setup the unit can be
configured as 2- or 8-channel device. In most cases 2-channel will be the better choice.
USB can be manually overridden by selecting a different mode. This will change the I/O routing,
but not disable USB itself. All inputs are still available via USB, outputs 3/4 can playback USB. In
all modes except Basic Mode DAC output PH 3/4 is freely configurable, any possible source can
be monitored independently from outputs 1/2.
The unit remembers all settings, and loads these automatically when it is switched on again. To
switch off, press the Standby button for at least 0.5 seconds.
5.3 Operation at the Unit
Useful information for a smooth start:
Turning the big VOLUME knob brings up the Volume
screen of the currently selected output. Pushing the big
Volume knob changes the volume setting between Outputs
1/2 and 3/4. The status bar at the bottom of the display
shows the current dB value of both volume settings. A white
rectangle around it (marker) indicates which volume the big
encoder is currently set to control.
The header shows the currently selected output as well as
hardware reference level (Ref Lev, see chapter 12.2.1).
The EQ is set up either directly within the I/O menu structure (key I/O, turn encoder 2 to access
Settings, Parametric EQ, Bass/Treble and Loudness), or via the graphical EQ screen which is
shown after pressing the EQ key. In this screen the cursor has three positions: top*, the EQ
parameter line below the response curve, and filter selection type for band 1 and 5 (Peak, Shelf,
Low/High Cut). The cursor is moved up or down by pushing encoder 1 or 2. When in the EQ
parameter line, all values are no longer grayed out. In this state the big encoder steers Gain,
encoder 1 Frequency, and encoder 2 Quality factor. This way the EQ is extremely fast set up
and edited.
The graph has 5 different colors matching the 5 bands that can be adjusted. If the line is just
grey the EQ is disabled (bypass). The EQ can be enabled in the second menu which comes up
by pushing the EQ key a second time.
*(shown by a 1 beside the current channel. Turning encoder 1 will change to the EQ settings of the other channels)
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The unit has several informative screens on the top level. These are Global Level Meters, Ana-
lyzer Input, Analyzer Output 1/2, Analyzer Output 3/4, State Overview and Dark Volume.
Change between them by pushing encoder 1 or 2 whenever any of them is displayed. To quickly
call them up simply press any of the 4 buttons several times.
In all these screens turning encoder 1 and 2 brings up the quick access to Bass and Treble, with
± 6 dB maximum boost/cut.
5.4 Overview Menu Structure
1 means upper small encoder (B),
2 lower small encoder (T).
Rotate moves horizontally, press vertically
(1 up, 2 down) through the menu structure.
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User’s Guide ADI-2 Pro FS R - v 3.3
5.5 Playback
In the audio application being used, the ADI-2 Pro must be selected as output device. It can
often be found in the Options, Preferences or Settings menus, as Playback Device, Audio De-
vices, Audio etc. After selecting a device, audio data is sent to an analog or digital port, depending on which has been selected as playback device.
Increasing the number and/or size of audio buffers may prevent the audio signal from breaking
up, but also increases latency i.e. output is delayed.
5.6 Analog Recording
For recordings via the analog inputs the corresponding record device has to be chosen.
Channels 1/2 of the ADI-2 Pro have digitally controlled gain and four hardware-based reference
levels. The digitally controlled gain offers a gain setting in steps of 0.5 dB within a range of 0 dB
to +6 dB, for fine-tuning the input sensitivity. The four hardware-based Reference Levels allow a
coarse adaption to the current source signal. The ADI-2 Pro has global as well as channel level
meters. Setting the correct Ref Level to avoid clipping/overload is easy to do.
The combo XLR/TRS sockets are designed for line signals. Sources that require higher input
impedances, like guitars, need an additional impedance buffer in front of the ADI-2 Pro.
5.7 Digital Recording
The easiest way to perform digital recordings with the ADI-2 Pro is to set the SRC to the currently used input (SPDIF or AES), then set the Clock to INT(ernal) and the desired sample rate –
then start recording.
The SRC serves as clock decoupler. When not using the SRC, the ADI-2 Pro must be in total
Sync to the external digital device, as either master or slave. Taking this into account, RME added a comprehensive I/O signal status display to the ADI-2 Pro, showing sample frequency, lock
and sync status in the State Overview screen and the bottom status bar.
The sample frequency shown in the State Overview screen is useful as a quick display of the
current configuration of the unit and the connected external equipment. If no sample frequency
is recognized, it will show - - (No Lock).
This way, configuring any suitable audio application for digital recording is simple. After connecting it the ADI-2 Pro displays the internal and external sample rate. This parameter can then be
changed in the application’s audio attributes (or similar) dialog.
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6. Power Supply
In order to make operating the ADI-2 Pro as flexible as possible, the unit has a universal DC
input socket, accepting voltages from 9.5 Volts up to 15 Volts. An internal switching regulator of
the latest technology with high efficiency (> 90%) prevents internal hum noise by operating
above audible frequencies. Internally the switching regulator is followed by standard linear regulators, followed by super low-noise linear regulators. Therefore the ADI-2 Pro achieves its technical specs even with less optimal power supplies. Or in other words: the choice of power supply
is not critical.
Still the unit includes a high-quality switching power supply, 12 V / 2 A, which not only accepts
any mains voltage between 100 V and 240 V (usable world-wide), but is also fully regulated
against voltage fluctuations and suppresses line noise. Additionally it only weights 150 g in spite
of its high power of 24 Watts.
The DC input of the ADI-2 Pro also allows for the use of a rechargeable lead-battery or LiPo
instead of a power supply, for completely independent mobile operation and ground isolation. A
matching connection cable (power jack 5.5 x 2.1 mm to terminals 6.3 mm) should be available
from various sources. Special power banks in the range of 10,000 mAh and up can be found
equipped with a 12 V output. These offer a perfect solution for mobility as well as ground isolated operation, for small money.
7. Firmware Update
The ADI-2 Pro might receive improved features or bug fixes by a firmware update. This update
will be available from the RME website, section Downloads, USB. Download the tool that matches your operating system (Mac or Windows), then unpack the zipped archive.
The Flash Update Tool updates the
firmware of the ADI-2 Pro to the latest
version. Under Windows it requires an
already installed MADIface series driver, which is found on the same download page.
Start the Flash Update Tool. It displays
the current revision of the ADI-2 Pro
firmware, and whether it needs an
update or not. If so, then simply press
the 'Update' button. A progress bar will
indicate when the flash process is finished (Verify Ok).
After the update the ADI-2 Pro needs
to be reset. Push the standby button to
switch it off for 5 seconds.
When the update unexpectedly fails
(status: failure), the unit's Safety BIOS
will be used from the next boot on, the
unit stays fully functional. The flash process should then be tried again.
The flash process does not affect user data like sample rate choices, EQ settings or Setups.
Returning to Factory State
In case a total reset is desired: hold encoder 1 and the VOL button pushed while turning on the
unit. This will reset all memory to factory default. Setups and EQ Presets stored by the user will
not be deleted.
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8. Features Explained
8.1 Extreme Power Headphone Outputs
During the development of the ADI-2 Pro an extensive research on today’s headphone amp
technology as well as headphones has been carried out. Many (many!) headphones later a maximum output level of +22 dBu (10 Volt) was set as development goal, as it will drive even insensitive headphones sufficiently, while a maximum output current of around 260 mA per channel
will result in lots of power for lower impedance phones (1.5 Watts @ 32 Ohm).
Limiting the current makes a lot of sense. It is needed to control the internal power supply, to not
fully overdrive (and destroy) weaker headphones, and prevents malfunction at short-circuit state.
The Extreme Power output stage acts like a small power amplifier, so it got a similar feature set:
a relay that mutes and interrupts the connection to the phones, a DC sensing circuit to prevent
DC at the output (DC destroys your precious phones already when the rated watts are not even
closely reached!), and an over-current protection circuit that will notice when a short-circuit
causes too high current, preventing the output stage to get destroyed. In light of the malicious
treatment of this output stage during development it needs to be mentioned that it can not be
destroyed by a short-circuit at the output nor by overheating. Still some extra safety won’t hurt,
and the over-current protection circuit got in.
A goal during development was to build a headphone amp that not only reaches very low THD
un-loaded (the typical way of measuring it), but very low THD values with a 32 or 16 ohm realworld load. This was achieved in the new Extreme Power headphone driver output stage. It uses
6-fold spread power technology, improved thermal conductivity and a special super-low distortion driver design. The result is THD below -110 dB at 32 Ohm load even near full output level
(clipping), the same SNR as the DAC provides (120 dBA), an output impedance of only 0.1
Ohms, totally stable operation, and a frequency response from 0 Hz up to 80 kHz, with just 0.5
dB decrease at the top end. The result: No audible hum, noise or distortion, fully transparent and
crystal clear sound at any volume setting, for any personal taste in any application.
And there is more. The headphone sockets of the ADI-2 Pro have sensor contacts. The unit
always knows when a headphone jack is inserted or removed. The DSP uses this information for
several superior, partly never-seen before features. For example when inserting the headphone
jack into Ph 3/4, the ADI-2 Pro activates the mute relay after half a second, then the DSP ramps
up the volume slowly from lower level to the last used state. Comfortable? Luxurious? Yes, but
the main reason for it was to give the user a chance to react. Extreme Power headphone outputs set to full output level, music already playing at full level, inserting the phones, and the moment the relay switches on the doctor is called, diagnosing sudden deafness - this should and
can not happen with the ADI-2 Pro. When the volume is ramped up one has the time to either
quickly set the phones off, unplug the phones again, or to grab the Volume knob to quickly turn it
down.
To guarantee that the Volume knob will be set to control the correct outputs in that moment, the
DSP also sets the Volume knob automatically to the output where phones had been plugged in.
And even returns the setting when the phones are unplugged again.
This is just an example of how intelligent and elaborate the control logic of the ADI-2 Pro has
been implemented. There are lots of such functions and features that might even stay unnoticed, making the unit behave fail-proof as well as easy to operate.
But isn’t +22 dBu, or Hi-Power as it is called in the menu, much too loud for modern phones?
That depends. There are still phones that need higher levels. Music can be low in volume but
consume a lot of power, especially with lots of sub-bass. And lots of headroom is always nice to
have. Typically with Hi-Power off, which equals +7 dBu maximum output level, modern music
and modern headphones, Hi-Power is mostly not needed. But you will notice that even with HiPower active, which requires to use a volume setting 15 dB lower as usual, the sound stays the
same, and there is no audible noise or hum at the phones output (provided the source is clean,
of course). So even at a Volume setting of -40 dB the ADI-2 Pro delivers perfect sound quality,
being a no-brainer in daily use as how to set it.
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8.2 Dual Phones Outputs
Many features and design decisions on the ADI-2 Pro come from personal usage and experience. For example when comparing headphones: it turns out to be very difficult when having just
one headphone output. Changing the phones on the head is already a disrupting process which
hinders easy comparison, but without proper level adjustment first, and the need to unplug one
and to plug the other, comparisons are only possible for coarse differences. At RME we are
used to compare headphones connected to a Fireface UFX or 802. These exceptional audio
interfaces have two independent phones outputs. The included TotalMix FX, a DSP based mixing engine, allows to route the same audio signal to both outputs, with individual vol ume settings,
and no need to unplug / plug anything. So if one phone is too low in volume it is simple to raise
it, or lower the other one, to get them on the same volume, making a comparison much easier.
The ADI-2 Pro has two stereo DA-converters to similarly provide two independent and individual
phones outputs. Adding a third DAC for the line outputs would raise cost, space and effort tremendously, while listening on two phones at the same time or comparing phones this way is a
seldom task. Therefore one phones output, labelled PH 1/2 on the front, shares the main (rear)
output signal. Although this phones output reaches the same technical specs as PH 3/4, and
also has the exact same Extreme Power output stage, it is considered the ‘spare’ Phones output
for comparing phones, dual phones usage, and balanced phones operation – or just use it as
another unbalanced line output. The main Phones output, which is independent from the rear
outputs, is PH 3/4. For most users it will be the only output ever needed and used. And because
it is the most often used one it was intentionally moved away from the Volume knob to ease
operation, resulting in an unusual arrangement with PH 3/4 left and PH 1/2 right.
As explained a major reason to have not only two, but two independent phones outputs is that it
offers a much better way to compare headphones. But there is more to it, see next chapter.
8.3 5-band Parametric EQ (PEQ)
Comparing headphones with the Fireface UFX and 802 comes with another, big advantage:
TotalMix FX controls a 3-band parametric equalizer (PEQ), again independent for both outputs.
So if one phone has too much or too little bass, it’s easy to reduce or increase lower frequencies
so the phones become more similar. This makes it much easier to hear the basic, but finer differences in the phones sound signature.
Having worked extensively with this luxurious double output solution, there is no question why
the ADI-2 Pro got two fully independent, identical Extreme Power headphone outputs, and individual equalizers for both outputs. This is indeed the premium way to compare headphones
seriously as well as efficiently.
While no equalization as well as listening only straight linear has been a mantra for many years,
research has proven that no ears are identical, and that especially in near-field listening (with
phones) the biological differences alone make individual equalization mandatory. No two pairs of
ears hear the same thing, that’s a fact. Additionally personal taste makes people like different
sound signatures, which can easily be copied or made more similar (equalized…) on different
headphones using a good EQ. The advantages of using an EQ outweigh any alleged disadvantages - which so often turn out to be wrong at closer inspection.
Having used PEQ to linearize as well as to better meet personal taste with a variety of headphones, RME found 5 bands of parametric EQ to be the best balance between occupied DSP
resources and efficient sound treatment. While it is true that on some phones rebuilding an exact response curve needs more than 5 bands, one quickly realizes that very narrow peaks and
notches make no audible difference when compensating them. Their acoustical energy is too
low to get audible. Ignoring those narrow peaks/notches and only taking care of deviations that
require a quality factor of 3 or below, the 5 band parametric EQ turns into a very efficient tool
even for problematic phones.
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This is one of the many major features that can’t be found on any similar device: a high-quality
5-band parametric EQ, usable at up to 768 kHz sample rate, easy to set up and adjust, with a
graphical display showing the resulting curve, and multiple storage places including individual
naming. So whatever EQ setting you need, it is loaded and modified quickly. And there is not
only one, but three such EQs, separately for the analog inputs, and both stereo analog outputs.
On a related topic: These days many people suffer from hearing loss in varying degrees. No
matter if it is biological, from abuse or an accident - hearing impaired is a plague of modern
times. And – no surprise when thinking about it – it never affects both ears identically. The number of people having one sided hearing problems is huge, but they have learned to live with an
industry that totally ignores them. Although the solution is as simple as logical – have the EQ be
adjustable independently for left and right. Basically digital EQs are calculated this way, the
common controls are just for making it easier to operate. The ADI-2 Pro includes an option
called Dual EQ – a heaven send feature for many, for sure.
Of course a 5-band parametric EQ is also suitable for speaker and room correction, another
application where separate EQ left/right setups are necessary. Using the ADI-2 Pro as DAC for
the main monitors will benefit from this and all the other typical RME features available on all
analog I/Os: Phase and Mono in various options, Width and M/S Processing.
8.4 Bass / Treble
The simpler form of EQ has been the Bass and Treble controls as they are found on any ‘stand-
ard’ HiFi stereo amplifier. They easily and quickly allow to modify the sound to your personal
liking (more or less Bass, more or less Treble, obviously). An even more useful application is to
quickly change the amount of Bass / Treble in smaller quantities so that music compilations
don’t have one song making the cones fall out while another one makes you think that really
happened. Producers and mastering engineers not only have their own taste, they also sometimes fail in providing a mix that is on an average sound level compared to others. In that moment a quick turn on the two ADI-2 Pro’s small encoders will make the music sound perfect.
These Bass and Treble controls are limited to ± 6 dB. Everything exceeding such values should
be handled by the EQ, and/or calls for better speakers/phones. The corner frequency and quality
factor of Bass and Treble is user-adjustable in the display’s menu, making this feature even
more useful. Adapt it to meet your speakers/phones or your personal taste – it will greatly improve your pleasure in listening to music again.
8.5 Loudness
Another legacy of HiFi amplifiers: there has not been a single one missing a feature called
Loudness. It tries to address the changes in frequency-dependent hearing sensitivity over different volume levels. If one listens to music loud, then drops the level by at least 20 dB, sound loses punch and glitter. HiFi amps tried to fight this effect by adding more bass and treble the lower
the volume was set. Unfortunately that never worked as intended, and just became an additional
bass/treble booster. Reason: the manufacturer of the HiFi amp could not know what volume any
position of the volume knob equals at the customer’s home. Room size, room dampening and
efficiency of the used speakers are all unknown.
But the effect of loss in perceived sound exists (read about the Fletcher-Munson curves), and
can be easily reproduced with any serious gear by comparing normal volume and DIM state
(usually -20 dB). The ADI-2 Pro offers Loudness for both analog stereo outputs, and probably is
the first time that Loudness works as intended. The user can decide how much maximum gain
in Bass and Treble should occur at lower volume settings. The user also sets the Low Vol Reference, where maximum gain is achieved. After extensive tests a 20 dB range has been defined
as range for maximum gain to no gain while increasing volume. That seemed to be the perfect
definition of the range that needs to be addressed by Loudness.
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Here is an example on how it works: the user’s typical lowest level listening volume is at -35 dB
at the unit. This value is now set by the user as Low Vol Ref in the Loudness menu. Then Bass
and Treble Gain can be set between 0 and +10 dB. Default is +7 dB for both. Increasing the
volume by turning the Volume knob causes the gain in Bass and Treble to be lowered smoothly
over a range of 20 dB. So when Volume is set to -15 dB, the music is not only quite loud, but
Loudness’ Bass and Treble are then at 0 dB gain. See chapter 34.15 for graphs.
No matter how sensitive the connected phones or speakers are, no matter how much increase
in Bass and Treble are desired – with the ADI-2 Pro one can finally adjust it to meet the personal
hearing and taste. Loudness finally works as it should have worked from the start - another
unique feature in the ADI-2 Pro.
8.6 SRC (Sample Rate Conversion)
The ADI-2 Pro includes an asynchronous stereo sample rate converter (SRC). A SRC allows a
conversion of the sample rate in real-time. The converter used in the ADI-2 Pro operates practically without loss of signal quality, so no audible artefacts or noise is added. It also handles intersample peaks (ISP) up to +3 dBFS without added distortion. In fact, the SRC works so well
that we could recommend to just leave it on at all times, thus eliminating all clock problems right
from the start. Which is the case for the SPDIF input in Auto mode.
The SRC offers a maximum conversion rate of 1:7 or 7:1, respectively. Thus, 192 kHz can be
converted to any sample rate down to 44.1 kHz, and 32 kHz can be converted to any frequency
up to 192 kHz. Higher sample rates than 192 kHz are not supported.
An SRC not only converts sample rates, it also serves as a clock decoupler. With SRC active,
even non-synchronizable devices (CD-players, DAT machines, etc.) can be used in a setup of
digital devices, just as if they were externally synchronized. The SRC decouples input and output
clock and sets the output clock to the common reference, thus allowing the combination of different clock-sources. For example having the ADI-2 Pro synchronized to an AES signal, a CD
player connected to SPDIF input can only be used when the SRC is set to SPDIF. It then decouples the clock of the non-synchronizable CD player, preventing clock problems and drop outs.
As the incoming clock phase is no longer fixed when the SRC is activated, the SPDIF Sync state
in the State Overview screen will always show lock.
When using the internal clock, every SRC also works as a jitter killer. However, the ADI-2 Pro is
equipped with SteadyClock FS, thus operating as perfect jitter killer with any clock source. However again, a jittery input signal might degrade the quality of the sample rate conversion. The
ADI-2 Pro therefore has a second SteadyClock exclusively for the current SRC input signal to
make the sample rate conversion process as reliable and transparent as possible.
A SRC can also be used to upsample audio. A 44.1 kHz source can be converted to 192 kHz in
real-time and thus played back with the DAC set to 192 kHz. The usefulness of this process is
questionable. There is zero content added, so the exact same audio is played back. The only
change is that the DAC's oversampling filters are moved far out of the audible range. But even at
44.1 kHz the ADI-2 Pro’s filters are inaudibly high, and the process of sample rate conversion
also uses those lower filters during its first conversion process.
8.7 Crossfeed
While headphones open the sound stage and make everything easier to hear and to locate by
spreading the narrow sound field of stereo speakers to the left/right extreme, some people
would like to have a listening situation that is more comparable to a standard speaker setup. The
ADI-2 Pro includes Crossfeed to address this wish. Crossfeed reduces the artificial surround
ambience that some productions have to make them sound better on speakers, but which
sounds unnatural on a headphone. It uses the Bauer Binaural method, with five selectable
strengths of narrowing the upper frequencies. This advanced method, which also includes a
small delay and correction of the frequency response, works quite well, and is another useful
addition as well as a unique feature on a device like the ADI-2 Pro.
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Details on internal settings
The Crossfeed effect is mainly defined by the filter frequency and the amount of crossfeed, here
given as damping factor:
1: 650 Hz, -13 dB (just a touch)
2: 650 Hz, -9.5 dB (Jan Meier emulation)
3: 700 Hz, -6 dB (Chu Moy emulation)
4: 700 Hz, -4.5 dB (30° 3 meter emulation)
5: 700 Hz, -3 dB (example how even stronger would sound)
8.8 DSP Limitations
There is never enough DSP power – no matter how much you add (frustrated developer).
That is true even for the ADI-2 Pro. Although being equipped with a quite capable 2.17 Giga
FLOPS DSP chip, plus using the FPGA to perform further calculations (RME’s virtual DSP for
mixing/routing, level meters, filtering, Crossfeed), 768 kHz sample rate takes its toll. The calculation power available at 48 kHz is divided by 16 (!) then. Even at 384 kHz it is just 1/8 of that at 48
kHz. The DSP in the ADI- 2 Pro performs:
Bass/Treble and Loudness for 6 channels
5-band parametric EQ for 6 channels
Standard phase functions for 6 channels
Crossfeed for 4 channels
30-Band bi-quad bandpass filter spectral analyzer
Peak Level meters for all channels
Display rendering
Volume control on 4 channels
Several controller-like functions, like volume ramp-up, mute, signal routing control etc.
Balanced Phones mode control
DSD to PCM conversion (for level meters)
At 48 kHz that is no big deal, at 192 kHz it already needs efficient coding and a better DSP chip.
But at 768 kHz you need a DSP with 4 times the power of the ‘better’ one. Therefore there is no
way around disabling some functions at higher sample rates. Fortunately those limitations have
only small impact in real-world usage:
At sample rates 352.8 kHz and up the Bass, Treble and Loudness function is deactivated.
The number of available EQ channels is reduced to 2 (1 x stereo). EQ can still be used with
Analog Input, Main Output 1/2 or Phones Out 3/4, but only one of these.
At sample rates 705.6 kHz and up Crossfeed or EQ (1 x stereo) can be active, not both at
the same time.
The high sample rates available in the ADI-2 Pro also exceed the capabilites of the digital I/Os.
Both AES and SPDIF are limited to 192 kHz, and there is no way around it (except a special,
one channel SMUX mode, see chapter 14.1.2, Setup Clock). Therefore all higher sample rates
are only usable analog and in USB mode. And in iOS mode when using an iPad/iPhone with an
app that supports such high sample rates (Neutron, Onkyo HF-Player etc.).
DSD can not be processed digitally. Therefore DSP functions like Bass, Treble, Loudness, EQ
etc. are disabled in DSD mode. The volume control is no longer done by the DSP, but the DAC
chip. You won’t notice that, volume operation at the ADI-2 Pro is seamless and behaves identical in any mode. In DSD mode the DSP performs an additional DSD to PCM conversion, to be
able to show the audio signal on the level meters and for the Analyzer – another unique feature
of the ADI-2 Pro.
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User’s Guide
ADI-2 Pro FS R
Basic and Stand-Alone Operation Details
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9. RME Multi-Remote Control (MRC)
The included infrared remote control, manufactured exclusively for RME, offers an almost complete remote control in a clear way. It is pre-set to control the ADI-2 DAC. Pressing a button
therefore causes the SEL(ect) LED to light up green. To control the ADI-2 Pro FS R code table 2
needs to be activated so that the SEL LED lights up in orange color. If the LED shows green, red
or blue color, another code table is active that is not compatible with the ADI-2 Pro FS R.
To change the code table:
Keep SEL pressed until the LED lights up constantly.
Press button 2. The LED turns orange and then off. The remote control now works with the
ADI-2 Pro FS R.
Note: SETUP – Options - Remap Keys/Diagnosis - Remap Keys must be set to Remote.
9.1 Keys and Functions
SEL. Choice between 4 different code tables. The ADI-2 Pro FS
R uses table 2, recognizable by the LED color orange.
Power On/Off. Press and hold the button for 1 second to switch
off.
B/T. Bass/Treble active or off (linear).
EQ. PEQ active or off (linear).
LD. Loudness active or off.
B+ / - . Increase or decrease Bass, same as encoder 1.
+ (VOL) - . Increase or decrease volume.
VOL. Corresponds to pressing the VOLUME button on the unit.
Short press results in mute, long press results in output change
when configured via SETUP - Options - Phones / DSD - Mute
Line - Toggle Ph/Line.
T+ / - . Increase or decrease Treble, same as encoder 2.
L / R. Change balance between left and right.
1, 2, 3, 4, 5, 6, 7. Remap Function Keys. In the SETUP – Options - Remap Keys/Diagnosis
menu, 52 different commands/actions are available for these seven programmable keys (see
chapter 14.1.1). Functions such as polarity, crossfeed and DAC filters can be controlled live from
the preferred listening position. The default settings are: 1 Source Analog, 2 Source AES, 3 AutoDark, 4 DIM, 5 Source SPDIF coaxial, 6 Source Optical, 7 Source USB. All Source keys require the Basic Mode DAC to become active.
Mute. Mute the audio output.
Since the ADI-2 Pro has two independent stereo outputs, most keys only influence the currently
selected or active output (see marker in the status bar).
The remote requires a standard lithium battery CR2025 (included). For first time usage please
tear out the transparent plastic isolation completely.
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9.2 Other Remote Controls
The ADI-2 Pro FS R can also be remotely controlled using third-party remote controls and custom IR transmitters. The well-known manufacturer Logitech has added the ADI-2 Pro to its remote control database. Others can use the codes documented here:
http://www.rme-audio.de/downloads/adi2pro_ir_commands.zip
Available are the original keys of the MRC and all 52 Remap commands with direct access
(without remapping).
10. Front Panel Controls
General operation and usage of the ADI-2 Pro are explained in chapter 5.2, Quick Start, and
chapter 5.3, Operation at the unit.
The ADI-2 Pro ships with Basic Mode Auto activated. In this mode the unit will automatically
reconfigure itself depending on the connected source, offering quick, easy and intuitive operation:
No digital input, no USB => Preamp mode
Digital input signal => AD/DA converter mode
USB connected => USB mode (USB interface operation)
These modes are explained in detail in chapter 17. The current mode is also shown for 2 seconds (Info Message) whenever it changes, and one time after power-on.
State Overview is especially useful to check the state of digital input signals as well as the current settings with USB. It also displays several warning messages which might explain why currently no sound is audible. See chapter 15.3 for details.
The following chapters explain all the controls and menu items in detail.
10.1 Keys
The four back-lit keys offer quick access of important parameters within the menu structure.
After pressing one of the four keys the corresponding menu is shown in the display. The unit
remembers the last selection per key, so re-visiting a formerly changed parameter is easy. To
leave the menu push the same key a second time, or any other key two times. The display will
revert to the level meter screen that was active before entering the menu.
10.2 Encoders
The encoders can be turned endlessly, but also pressed, adding a push button function. The
current functionality of all encoders is shown in the display. The big Volume knob usually controls volume for outputs 1/2 or 3/4. The current assignment is indicated in the display’s status
bar by a marker around the volume value.
Turning the small encoders 1 and 2 either changes the current parameter, or moves the selection/cursor horizontally to the next page. Pressing the encoders 1 and 2 moves the selection/cursor vertically, up with 1 and down with 2, as indicated by the arrows in the display.
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Example: Press the key SETUP. The menu Setups is now
shown. 1 within the circle on the right side indicates that by
turning encoder 1 more pages are available. Turn encoder 1
left to enter Options. Now turn encoder 2 to scroll horizontally through all the subpages offered under Options: SPDIF /
Remap Keys, Device Mode / DSD, Clock. By pressing encoder 2 the cursor moves down, by pressing encoder 1 back
up. On a selected field or entry, 2 to the right indicates that
the current parameter can be changed by turning encoder 2.
Change Clock Source and Sample Rate to see how easy it
is to select and change important settings.
11. VOL
The VOL key brings up an extended volume screen with
balance control. Pushing the big Volume knob changes the
volume setting between outputs 1/2 and 3/4, which can
then be adjusted by both Volume knob and encoder 1. Encoder 2 sets the Balance parameter. Turning encoder 2
sets the Balance parameter. A push on encoder 2 (T) sets
Balance to center (<C>).
The volume and balance setting is also found in the menu
I/O, Settings, at the end of the list.
The status bar at the bottom of the display shows the current dB value of both volume settings.
A white rectangle around it (marker) indicates which volume the big encoder is currently set to
control.
A push on encoder 1 (B) mutes the current output. The text in the blue field shows Main 1/2 -
muted. A second push exits the mute state.
Pushing the VOL key a second time enters the Dual Volume screen, showing both volume settings at the same
time. Encoder 1 controls Volume 1/2, encoder 2 Volume
3/4, and the big Volume knob both. This allows to set the
outputs at individual levels, but also to control those simultaneously. The linked control operates on a relative base,
with individual volumes staying intact when increased or
decreased, even to maximum and minimum.
In the Dual Volume screen, both outputs can be muted by pushing encoder 1 (B) and encoder 2
(T) respectively.
Pushing the VOL key a third time reverts to the level meter screen that has been active before.
Note: The Dual Volume screen is not available in Balanced Phones mode. Main Out defaults to
Auto Ref Level enabled. The current volume setting is then shown as dBr (dB relative).
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12. I/O
The I/O menu has all the settings for the three analog stereo I/Os Analog Input, Main Output 1/2
and Phones Out 3/4. The submenu Parametric EQ mirrors the settings done in the graphical EQ
screen. The submenus Bass/Treble and Loudness as well as some phase functions are only
found on the two analog stereo outputs.
12.1 Analog Input
12.1.1 Settings
Subpage Settings has the following entries:
Ref Level
Sets the reference level for the analog inputs 1/2. Choices
are +4 dBu, +13 dBu, +19 dBu, +24 dBu, referenced to
digital full scale level (0 dBFS).
Auto Ref Level
ON or OFF. Default: OFF. In case of overload Auto Ref
Level will switch the Ref Level to the next higher setting.
This process is repeated until +24 dBu is reached. In case
Trim Gain was active it will be set to 0 dB first.
Trim Gain Left, Trim Gain Right
Digital amplification of the input signal between 0 and +6 dB, in steps of 0.5 dB. Main use is to
fine-tune the input sensitivity so that it matches the reference output level of external gear.
Phase Invert
Available settings are Off, Both, Left and Right. Inverts the phase (180°) on the corresponding
channel.
M/S-Proc
Activates M/S processing. Monaural content is sent to the left, stereo to the right channel.
AD Filter
Short Delay Sharp, Short Delay Slow, Sharp, Slow. The analog to digital conversion can be done
using four different filters. Default is SD Sharp, offering the widest and most linear frequency
response and lowest latency. SD Slow causes a small drop in the higher frequency range, but
offers a less aggressive (less steep) filter. Sharp and Slow are FIR filters with different impulse
responses. See the Technical Reference section for graphs illustrating the results in frequency
response and impulse response.
Note: At sample rates higher than 192 kHz DA Filter selection is no longer available. The ADC
then uses a fixed Slow filter.
Dual EQ
OFF or ON. Default: OFF. When set to ON, the 5-band parametric equalizer can be set individually for left and right channel.
AD Conversion
PCM or DSD. Default: PCM. DSD will not become active at sample rates below 176.4 kHz.
When selecting DSD the current DSD rate is shown as well. It changes with the chosen sample
rate (SETUP - Options - Clock).
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12.1.2 Parametric EQ
Subpage Parametric EQ has the following entries:
EQ Enable
ON, OFF. Default: OFF.
Band 1 Type
Available settings are Peak, Shelf, High Cut and High Pass
(Low Cut). All filters are adjustable from 20 Hz to 20 kHz, at
a Q of 0.5 to 9.9. Cut/Pass have a fixed 12 dB/oct filter
steepness.
Band 2-4 Type
Not available, fixed to Peak.
Band 5 Type
Available settings are Peak, Shelf or High Cut. High Cut is adjustable from 200 Hz to 20 kHz, at
a Q of 0.5 to 5.0 and a fixed 12 dB/oct.
Band 1-5 Gain
Available settings are -12 to +12 dB in steps of 0.5 dB.
Band 1-5 Frequency
Adjustable from 20 Hz (200 Hz bands 4/5) to 20.0 kHz, in steps between 1 Hz and 100 Hz.
Band 1-5 Q
Quality factor is adjustable from 0.5 to 9.9 in bands 1 to 3, and 0.5 to 5.0 in bands 4 and 5, in
steps of 0.1 dB. This equals a bandwidth setting of 2.54 (0.5), 0.29 (5.0) and 0.146 (9.9).
Subpage Parametric EQ R is only shown with Dual EQ set to On. It has the exact same
entries as listed above.
12.2 Main Output 1/2
12.2.1 Settings
Subpage Settings has the same settings as listed for Analog Input, plus:
AD/DA Source
The source of the Main Output 1/2 signal is automatically selected based on the current mode:
Preamp mode: Analog inputs 1/2
USB: Playback channels 1/2
Dig Thru mode: current digital input signal
AD/DA converter: current digital input signal
DAC: current digital input signal and clock
The entry AD/DA Source is therefore usually grayed out.
Only in AD/DA and DAC mode the input signal can be chosen between Auto, SPDIF, AES and Analog. This allows to
choose between all currently attached digital input signals
for conversion to output 1/2.
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Ref Level
Sets the reference level for the analog outputs 1/2. Choices are +4 dBu, +13 dBu, +19 dBu, +24
dBu, referenced to digital full scale level (0 dBFS). This setting is also valid for the front output
PH 1/2, with PH 1/2 having 3 dB higher output level. This way the setting +4 dBu becomes +7
dBu output level, +19 dBu becomes +22 dBu at the phones jack. These two settings are therefore identical to Hi-Power Off and On at Phones Output 3/4.
Auto Ref Level
ON, Off. Default: ON. See chapter 21.3 for details.
Mono
OFF, ON, to Left. Default: OFF. The option to Left sends the sum of left and right channel to the
left output only.
Width
Defines the stereo width. 1.00 equals full stereo, 0.00 mono, -1.00 swapped channels.
Crossfeed
OFF, 1, 2, 3, 4, 5. The Bauer stereo to Binaural crossfeed effect emulates speaker playback by
reducing the stereo width in the treble range. Adjustable in five steps.
DA Filter
Short Delay Sharp, Short Delay Slow, Sharp, NOS, Short Delay Low Dispersion. The Digital to
Analog Converter chip offers several oversampling filters. Default is SD Sharp, offering the widest and most linear frequency response and lowest latency. SD Slow causes a small drop in the
higher frequency range, but has a less aggressive (less steep) filter. Sharp and Slow are similar,
but have a higher latency. NOS is the filter with the smallest steepness and therefore affecting
treble more than the others, but offers the best impulse response. See the Technical Reference
section for graphs illustrating the results in frequency response and impulse response.
Note: NOS deactivates the option De-Emphasis. At sample rates higher than 192 kHz DA Filter
selection is no longer available. The DAC then uses a fixed Slow filter.
De-Emphasis
Auto, OFF, ON. Default: Auto. For manually de-/activating the DAC's de-emphasis filter. See
chapter 34.5.
Volume
Mirrors the direct volume control via Volume knob or encoder 1. The output level can be set
between -96 dB and +6 dB, mostly in steps of 0.5 dB. The encoders use a special accelerator
algorithm. Turning the knob fast increases the step size. At moderate turning speed the changes
in dB follow the intended volume change. Only at slower turning the finest steps will be used.
Lock Volume
Deactivates volume control via the big VOLUME knob. Volume within the menu still works, and
is used to set the desired output level. Active Lock is indicated in the VOL and Volume screen
and the status bar.
Balance
Mirrors the balance control in the VOL screen. Adjustable from L 100 (left) through <C> (center)
to R100 (right).A quick turn jumps from L or R to <C> and vice versa.
Mute
Mutes the output. Can also be controlled via the VOL screen and Remap Function Keys.
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12.2.2 Bass/Treble
Subpage Bass/Treble has the following entries:
B/T Enable
OFF, ON. Default: ON
Bass Gain
Current Bass amplification for the current channels as set by encoder 1 (B). Adjustable between
-6 dB and +6 dB in steps of 0.5 dB.
Bass Freq
Corner frequency of the shelf bass filter. Adjustable from
20 Hz to 150 Hz in steps of 1 Hz. Default: 85 Hz.
Bass Q
The quality factor of the filter is adjustable from 0.5 to 1.5.
Default 0.9.
Treble Gain
Current Treble amplification for the current channels as set by encoder 2 (T). Adjustable between -6 dB and +6 dB in steps of 0.5 dB.
Treble Freq
Corner frequency of the shelf treble filter. Adjustable from 3 kHz to 10 kHz in steps of 100 Hz.
Default: 6.5 kHz.
Treble Q
The quality factor of the filter is adjustable from 0.5 to 1.5. Default 0.7.
12.2.3 Loudness
Subpage Loudness has the following entries:
Enable
ON, OFF. Default: OFF.
Bass Gain
Maximum Bass amplification. Adjustable between +1 dB
and +10 dB in steps of 0.5 dB. Default: +7 dB
Treble Gain
Maximum Treble amplification. Adjustable between +1 dB
and +10 dB in steps of 0.5 dB. Default: +7 dB
Low Vol Ref
Reference level for highest Bass/Treble amplification, referenced to the Volume set in dB. Available range is -90 dB to -20 dB. Default: -30 dB. A volume setting below this point will have maximum Bass/Treble gain, all volume settings above this point will have lower Bass/Treble gain. 20
dB above the Low Vol Ref setting the Bass/Treble gain will be zero.
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12.3 Phones Output 3/4
Subpage Settings has the same settings as listed for Main Output 1/2, plus:
Source
Default: Auto. The source of the output Phones Out 3/4 can
be chosen manually anytime. Available options are: Auto,
AES, SPDIF, Analog, USB 1/2, USB 3/4. Auto here not only
means current or available signal, but also channels 1/2.
Hi-Power
OFF, ON. Default: OFF. Reference level for 0 dBFS is +7
dBu at the output. With Hi-Power on reference level is 15
dB higher, +22 dBu.
Auto Ref Level
ON, OFF. Default: OFF. See chapter 21.3.
13. EQ
The key EQ brings up a graphical EQ screen (Bode plot) to
set the EQ quickly and with full overview. It is available on
all analog I/Os. The I/O - Settings submenu Parametric EQ
mirrors the settings done in this screen.
On the top level, turn encoder 1 to change between Analog
Input, Main Output 1/2 and Phones Out 3/4. Turning encoder 2 will scroll through all 5 bands, as can be seen in the
parameter line. This function allows to see/check/verify all
parameters of all bands quickly, without the danger of
changing any of them.
Push encoder 2 to move the cursor to the parameter line,
with all values shown in white color. It is now possible to
adjust all parameters by turning the three encoders. The
Volume knob changes Gain, encoder 1 Frequency, encoder
2 Q (Quality factor). All changes are shown in real-time as
frequency response curve, making it very easy to find the
desired settings.
To change to the next band push the encoder Volume.
The five bands have different colors to clearly show what is currently selected: band 1 red, band
2 yellow, band 3 green, band 4 light blue, band 5 dark blue.
Band 1 and 5 also allow to set the filter to Peak or Shelf mode, plus Hi Pass/Hi Cut. This func-
tion is accessed by pushing encoder 2 so that the cursor moves down to the small filter symbol
in the lower right. It is no longer grayed out. Turning encoder 2 now toggles through the available
settings, with the symbol changing according to the selected function.
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Another push on encoder 2 changes to the graphical EQ
Preset selection screen. Turning encoder 2 will scroll
through all available EQ presets with the frequency graphics
showing the respective curve, and the parameter line showing the preset name. In this screen Volume, volume selection and channel selection (encoder 1) are also available.
Notes
The frequency graphics give a precise overview of the filter
results. Overlapping filters influence each other. This can be used to achieve more than 12 dB
gain, or to generate difficult frequency response optimizations.
The ADI-2 Pro has an internal headroom of 24 dB. Extreme boosts with overlapping filters could
cause an internal overload. Such an overload will be visible as it is displayed by the level meter
below the EQ, as well as the channel’s level meter. Reducing the output volume will prevent any
clipping as long as the headroom of 24 dB is not exceeded. In real-world operation that is always
the case, the ADI-2 Pro will not distort internally.
The EQ can cause distortion for the digital outputs when used on the Analog Input. Again the
level meters will clearly show this error condition. Reduce the input’s sensitivity by selecting a
higher Reference Level in such a case.
If the frequency graph is shown as grey line the EQ is disabled. There are two ways to change this state:
Push key EQ again to change to the page EQ Enable /
Presets, see below.
Push key I/O, select current channels, Subpage Para-
metric EQ, EQ Enable ON or OFF
Pushing the EQ key a second time brings up the EQ Enable / Presets screen. In this screen the
EQ can be switched on and off, and EQ presets can be stored and loaded comfortably.
Use encoder 1 to change between the subpages Analog Input, Main Output 1/2 and Phones Out
3/4. These subpages have the following entries:
EQ Enable
Default: OFF. Options are ON, OFF, L, R (L and R are only available with Dual EQ activated).
Preset Select
Load or store up to 22 different EQ settings. The first choice,
Manual, holds the current, unsaved EQ settings. The second
choice, Temp, holds the settings of a loaded and then modified Preset. This scheme lets the user easily change and
compare three different EQ settings: the manual one, the
stored preset and the modified preset, without loosing
changes while listening to a different set of EQ settings.
The last entry (21, Clear) corresponds to the factory default with all bands at 0 dB. It is not available to store a preset, but is used for resetting one by overwriting. A preset reset in this way,
resulting in an 'empty' preset, is marked with (lin).
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The presets are independent from and not stored with Setups (see chapter 14.2). EQ Presets
are therefore always available, no matter which Setup has been loaded. The Setup does include
the current EQ setting, which on load is written into the memory slot Manual.
Name
Allows to edit the name of the current preset and to edit the
name during the store process. Turn encoder 2 to select a
letter, number or symbol, then press encoder 2 briefly to enter the next sign. After the last sign the cursor jumps to the
field Store to. The name can consist of up to 14 signs. Turning encoder 1 gives access to all existing preset names, so
copying and modifying a preset can be done more quickly.
Changing the name is always stored immediately during editing, without further confirmation.
Leaving this field the name is automatically adjusted to the right. Adding signs to the front and
rear is possible afterwards. A quick turn to the left brings up space, which is also used to quickly
delete letters. Available signs are:
Space, Aa to Zz, + - / ( ) * ; : . , ! # $ & < > = ' I @, 0 - 9
Save to
Use encoder 2 to select the slot where the current preset should be stored to. To store press
and hold encoder 2 for one second.
14. SETUP
The key Setup gives access to two top level screens: Options and Load/Store all Settings. Options has the subpages SPDIF / Remap Keys, Device Mode / DSD, Clock, Phones and Display.
14.1 Options
14.1.1 SPDIF / Remap Keys
Subpage SPDIF / Remap Keys has the following entries:
SPDIF In
Available settings are: Auto, Coax, Optical. Default: Auto.
SRC (Sample Rate Converter)
Available settings are: Off, AES In, SPDIF In. Default: SPDIF In. Note: In case a DoP signal
(DSD) is detected, SRC is automatically switched off.
SRC Gain dig.
Available settings: 0 dB, -3 dB. Default: 0 dB. The SRC uses digital headroom to handle intersample peaks up to +3 dBFS without distortion. With the -3 dB setting, an SRC signal with up to
+3 dBFS can be recorded via USB without distortion, or passed on at the AES/SPDIF output in
Dig Thru mode. A normal SRC input signal with 0 dBFS then has a level of -3 dBFS for these
two paths.
Optical Out
Available settings are: SPDIF, ADAT. While the input adapts to the received signal automatically, the output needs to be switched manually. In Dig Thru mode with an ADAT signal received,
the output is switched to ADAT automatically with all 8 input channels passed through.
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Remap Keys
OFF, ON, Remote. Default: Remote. Allows to assign 52 different functions/actions to the four
function keys on the device and seven programmable keys on the remote, configurable via the
following seven entries:
VOL Key, I/O Key, EQ Key, SETUP Key, IR Key 5-7.
Available functions/actions:
Setup 1 to 9, Mono 1/2, Mono 3/4, Mono 1/2 to L, Mono 3/4 to L, Mute 1/2, Mute 3/4, Mute all,
Loudness 1/2, Loudness 3/4, EQ In 1/2, EQ Out 1/2, EQ Out 3/4, B/T Out 1/2, B/T Out 3/4,
EQ+B/T+Ld 1/2, EQ+B/T+Ld 3/4, Toggle Ph/Line, EQ+B/T+Ld 1-4, Polarity, Crossfeed 1-5, DA
SD Sharp, DA SD Slow, DA Sharp, DA Slow, DA NOS, DA SD LD, Toggle View, DIM, Mono all,
Mono all to L, Loudness all, EQ Out all, BT all, AutoDark.
Additionally Source AES, Source SPDIF c, Source Optical, Source Analog, Source USB 1/2 are
available. In Basic Mode DAC (and only there), these realize a switching of the input to be monitored on the device and via the remote control (see chapter 34.23).
The original function of the key, entering the menu, is still available by pushing the key for half a
second.
Diagnostic Data, Test Results
Please ignore. For internal use only.
SW Version
Shows the current version number and date of the DSP firmware.
14.1.2 Clock
Clock Source
Choices are Auto, INT (Internal, Master), AES, SPDIF.
Sample Rate
Choices are 44.1, 48, 88.2, 96, 176.4, 192, 352.8, 384, 705.6
and 768 kHz.
At 352.8 or 384 kHz sample rate the level meter display shows a single channel of audio sent
out from SPDIF, or received in case of a signal of 192 kHz sample rate. The reason is a special
SMUX mode in the ADI-2 Pro. When run at 3xx kHz, the ADI-2 Pro will split the data of the left
analog input channel to the AES/SPDIF output channels left and right, at half the sample rate
each. Using measurement software like HpW Works, (www.hpw-works.com) which supports this
mode (2x speed), and any 192 kHz capable RME audio interface, enables single channel analog
measurements at 384 kHz sampling over SPDIF optical – with the device under test galvanically
isolated from the measuring system (interface/computer).
14.1.3 Device Mode / DSD
Basic Mode
Choices are Auto, AD/DA, USB, Preamp, Dig Thru and
DAC. See chapter 17.
CC-Mode
Choices are Stereo and Multi-channel. The ADI-2 Pro supports two Class Compliant modes: 2 channel I/O, which
allows to use sample rates up to 768 kHz even with iOS
devices, and 6/8 channel mode to give access to all I/Os
simultaneously. In multi-channel mode the sample rate is
limited to 192 kHz. To be able to change the mode USB
must be disconnected.
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Dig. Out Source
Default, Main Out. Copies the signal Main Out 1/2 (including EQ and volume) to the digital outputs AES and SPDIF/ADAT. Useful when connecting active monitors with digital inputs.
DSD Direct 1/2
OFF, ON. Default: OFF. When activated a DSD playback will use DSD Direct mode over the
rear outputs 1/2. As DSD Direct bypasses all DSP calculations and volume control, the only way
to change the output volume is by setting different reference levels. Therefore in DSD Direct
mode Phones outputs 1/2 are disabled.
DSD Filter
When DSD Direct mode is active, high-frequency noise filters reduce out-of-band noise, which
might have negative impact on other equipment. While 50 kHz is optimized for DSD64 and 150
kHz for DSD 128 and 256, the user can freely try both at any DSD rate.
DSD Detection
Default: ON. Option to deactivate the automatic DSD detection on SPDIF, AES and USB.
14.1.4 Phones
Dual Phones
OFF, ON. Default: OFF. With Dual Phones ON the phones output PH 1/2 will be active. Default
state is Off, as PH 3/4 is the main phones output and should be used exclusively unless two
phones are to be connected.
If Dual Phones is on and two phones are plugged in, a push on VOLUME toggles between 1/2,
3/4 and linked volume control (with the marker over both). When VOLUME is turned the Dual
Volume screen is shown then.
Bal Phones Mode
OFF, ON, Auto. Default: OFF. In Balanced Phones mode output PH 3/4 carries the left channel,
output PH 1/2 the right channel. See chapter 18 for details. When Auto is selected, the balanced
phones mode is automatically activated as soon as both Phones outputs detect a connector
being plugged in. This feature temporarily deactivates DSD Direct mode if active.
Note: When active, the rear analog outputs are muted automatically.
Phones <=> Line
OFF, 1/2, 3/4, 1/2+3/4. Default: OFF. Activates the ability to toggle mute between Phones Out
and rear Line Out. Pushing the VOLUME knob for half a second will then switch between loudspeakers connected to the rear and phones plugged into the front. This function can also be
controlled by one of the four function keys via Remap Function Keys.
Mute v. TRS 1/2
ON, OFF. Default: ON, but grayed out. As soon as a plug is detected in PH 1/2 the rear outputs
1/2 are muted. Note: This function requires Dual Phones to be on to become accessible. With
Mute v. TRS 1/2 ON the channels Phones 1/2 and Mains Out 1/2 have separated settings. Although both outputs alternately play back the same signal, all settings (Settings, EQ, BT) can be
different, and are separately stored in the background.
Mute v. TRS 3/4
ON, OFF. Default: ON. If set to ON a plug detected in PH 3/4 mutes the rear outputs 1/2.
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14.1.5 Display
Display Mode
Available settings are: Default, Dark. The dark scheme inverts the white background and black numbers/text to black
background and light-grey numbers/text.
Meter Color
Green, Cyan, Amber. Default: Green. Changes the color of
the level meters for PCM and DSD mode.
Hor. Meter
The horizontal stereo level meter below the Analyzer can show the peak level before all DSP
processing (Pre, equals the current input level from USB playback and SPDIF In), after all processing including volume control (Post), or both at the same time (Dual). The outer thin line is
the Pre level. In Dual mode the peak values to the right refer to Post level.
AutoDark Mode
OFF, 10s. Automatically turns off all LEDs and the display (exception: Standby button) after 10
seconds of user inactivity. A key press and turning an encoder will temporarily activate LEDs and
display again. Info and warning messages are shown for 3 seconds.
Show Vol. Screen
ON, OFF. Default: ON. When turning the VOLUME knob the Volume screen is shown.
LCD Brightness
Adjustable from 20% to 100%. Default is 80%.
LCD Tint Control
Adjustable from -8 (yellow) to 8 (blue). Enables compensation of the display's colour deviation as
well as suiting the user's taste.
14.2 Load/Store all Settings
This option allows to save the whole state of the unit as Setup in 9 memory slots. The EQ Presets are not included, they are stored separately and are available for any setup. The current
state of the EQ is also stored, and written to the EQ memory slot Manual during load of a Setup.
The page Setups, Load/Store all Settings, has the following entries:
Setup Select
Choices are Load 1-9, Factory (Reset All) and Store 1-9.
Name
Allows to edit the name of the Setup during the store process. To edit an existing name load the respective Setup
and store it on the same memory slot with edited name. See
EQ - Name for details about the Edit operation.
Start
Press 1s. Pressing and holding encoder 2 for at least one second triggers the action selected
(Load or Store).
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Returning to Factory State
In case a total reset is desired: hold encoder 1 and the VOL button pushed while turning on the
unit. This will reset all current settings to factory default. User-stored Setups and EQ presets are
not affected. The same action is performed by loading Factory via Setup Select. Note that the
reset will be incomplete when the unit is connected to USB while performing the reset.
When holding encoder 1, 2 and the VOL button pushed while turning on the unit, user-stored
Setups and EQ presets are still not affected, but their names are reset as well.
15. Top Screens
The ADI-2 Pro has four different meter screens: a global level meter that shows all signal levels
of all I/Os simultaneously, an Analyzer showing the audio signal content on the analog inputs
and both analog outputs 1/2 and 3/4, a state overview showing the digital states of AES, SPDIF
and USB, and a dark Volume screen with comprehensive information.
Pushing the small encoder 1 or 2 while any meter screen is active cycles through all meter
screens. To quickly call them up simply press any of the 4 buttons one or two times.
15.1 Global Level Meter
Shows the signal present at all inputs and outputs (except
USB). The upper labels refer to inputs Analog 1/2, AES and
SPDIF, then outputs Analog 1/2, Analog 3/4, AES and
SPDIF.
At sample rates above 192 kHz AES and SPDIF are grayed
out as they are no longer functional.
All level meters (vertical and horizontal) of the ADI-2 Pro are peak level meters with peak hold
function, band limited to 40 kHz even at higher sample rates. OVR (Over) are displayed in red
color at the upper end. For digital I/Os, Over appears when the signal is at the maximum possible level. As usual in digital audio, Over is already displayed even if there is no real overload.
OVR (Over) appears for the rear XLR and TRS outputs at +2.5 dBFS, since the ADI-2 Pro FS R
has this digital headroom. For the Phones outputs, OVR appears from +1 dBFS, since their analog circuitry limits the headroom here. In the horizontal meters of the EQ and Analyzer, the current peak hold value is displayed numerically on the right. Levels below -119 dB appear as UFL
(Underflow).
15.2 Analyzer
The Analyzer is based on RME’s famous Spectral Analyzer
in DIGICheck. It uses 29 biquad bandpass filters for high
separation between the bands, providing outstanding musical visualization. Using carefully selected attack and release
times the display is responsive, but still easy to read. On top
it uses RME’s own Max LR technique to prevent 6 dB high-
er level display for monaural signals, and zero display with
out–of-phase signals.
Thanks to the high-resolution IPS panel even smallest details are clear to see. Music content
analysis is possible even when viewed from a greater distance.
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The Analyzer operates at any sample rate, and even with DSD. There are no parameters to
change, and the shown frequency range is always the human audible range, 20 up to 20 kHz.
To be able to also show DC content the lowest band is not a band-pass filter, but a low pass,
catching the whole range from 0 Hz up to 30 Hz. With some unusual signals it therefore can
happen that the level shown will be a bit higher than expected.
As opposed to most other solutions no FFT (Fast Fourier Transform) is used. RME's Spectral
Analyzer performs a true band-pass filter calculation, as in professional hardware devices. The
frequency distance between the filters is scaled matching human hearing. The highly optimized
code allows to run a 30 band analyzer with 60 dB range, sharp filters and 0.5 dB steps accuracy
per band, on the ADI-2 Pro DSP, even at 768 kHz sample rate.
The most important application using a Spectral Analyzer is the visualization of frequencies and
levels found in music or speech. The Analyzer shows levels and frequencies even at the edge of
the human ear's abilities – or that of the used speakers and headphone. The visual display helps
to train ones ears, makes coarse errors visible, and shows what sometimes might stay unnoticed. For example many speakers won't let you hear frequencies below 30 Hz. Simply look on
the Analyzer to see what's going on in the underground.
15.3 State Overview
The State Overview screen is a typical RME feature. Since 20 years we prefer to give our cus-
tomers more information at hand than just ‘something there’. The Settings dialogs of our audio
interfaces include detailed Input Status analysis to simplify setup and ease trouble-shooting.
Additionally RME provides a free tool with any audio interface, DIGICheck, which analyzes levels, Channel Status and bit stream content, the true hardware sample rate and much more.
When the ADI-2 Pro is connected but no sound can be
heard then RME’s analysis tools come to the rescue. The
State Overview screen is there to track down the problem in
an easy to understand and efficient way. Although it might
look simple (fully intentional!), it includes detailed analysis
that outperforms any other similar device.
Shown are the current states of the digital inputs SPDIF
optical and coaxial, AES, USB connection and the audio
transmitted, and the SRC. The current Clock source is mentioned in full length, although this information is found abbreviated in the status bar at the bottom.
The status bar always shows the current volume setting of outputs 1/2 and 3/4, the currently
selected output for the Volume knob by a marker around the outputs, the current clock source,
and the current sample rate. In case of Sync problems the sample rate will either fluctuate or be
shown in red – or both. This information is available in nearly all screens, and helps to get a
quick overview of the current state. The State Overview screen now extends this information in
great detail.
The currently selected SPDIF input, manually or automatically, is shown as SP op or SP co
(optical or coaxial). The SYNC column shows No Lock, which equals no signal present, as - -.
And lock and sync, according to the current clock state of the respective input. Note that with
SRC active the respective input will always show lock, not sync, because the phase relation
between internal and external clock has no meaning anymore.
In case of USB conn (connected) is shown as soon as a valid USB connection is established.
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The column SR shows the hardware measured sample rate for the SPDIF and AES input. It will
even display values that can not be set at the ADI-2 Pro itself, for example 32, 64 and 128 kHz.
In case of USB the sample rate is not measured but set by the external computer or iOS device,
and can be verified here, up to the highest value of 768 kHz.
The State column shows the Channel Status, Consumer (cons) or Professional (pro), for incoming SPDIF and AES signals. In case a DoP (DSD over PCM) header is detected DSD is
shown. With USB the state column shows the current channel mode, 2/2 or 6/8, or DSD if a DoP
header is detected.
The Bit column shows the amount of bits found in the SPDIF and AES audio signal. Note that a
24 bit signal that is shown as 16 bit is indeed 16 bit, but a signal shown as 24 bit might contain
only 16 bit real audio plus 8 bits of noise…
But SPDIF and AES can also transport AC-3 and DTS encoded surround sound. This signal
sounds like chopped noise at full volume. Therefore the ADI-2 Pro receiver circuit checks the
Non-Audio flag within the Channel Status. If found the signal is already muted directly in the
receiver. An error message is shown in red colour in the lowest line, saying SPDIF NON-AUDIO,
explaining why there is no sound at the analog outputs despite a valid input signal.
Finally also Emphasis, a special treble boosting equalization method from the early days of digital audio, is detected and shown as SPDIF WARNING EMPHASIS. See chapter 34.5.
The inclusion of the SRC status helps to find simple errors that can happen due to the many
options of the ADI-2 Pro, which naturally reside in different parts of the menu structure. For example the SRC might be active, but SPDIF signal is dead audio-wise. A quick glance at the
State Overview will reveal that the SRC was set to work on the wrong input (AES)…
15.4 Dark Volume
This screen has been added as alternative to the other meter screens, which even with the dark theme selected might
be disturbing in some situations. Dark Volume shows the
current volume setting of both analog outputs, their current
reference level setting, the amount of Bass/Treble gain applied, and the Balanced Phones mode. It is low in brightness, still fully readable in bright environments, and has zero
flickering or moving elements (level meters). Volume and
Bass/Treble adjustments happen right in this screen, which
makes its functionality not only visually pleasing.
Despite this screen having no level meters, overloads caused by Volume, PEQ or Bass/Treble
are easy to recognize. The big gain numbers are coupled to the over detection of the analog
outputs. They change their color to red when overload occurs.
As with the other meter screens, once selected it becomes the default screen, automatically
shown after power-up or when leaving a menu.
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16. Warning Messages
The ADI-2 Pro will show different warning messages and provide guidance in certain cases.
Hi-Power Mode Active
When Hi-Power mode is active with the Volume set higher
than -15 dB and a phone is plugged in, this message reminds
the user to check the current volume setting, and to make
sure the used headphone will stand the high output power
without getting destroyed. The audio signal is held at a low
volume until the gain is set to -15 dB or below. The Volume
knob is active on the current phones output and can be used
to turn down the volume. Once -15 dB is reached the volume
is ramped up to the current gain.
Pressing encoder 1 removes the message immediately, with volume ramped up within 2 seconds to the set value.
The message will also vanish when the phone is unplugged again.
This message is not shown when Volume is set to -15 dB or lower, or when the device is
switched on while the phones are already connected.
Dual Phones Mode required
Phones output PH 1/2 is considered an additional output for
two specific cases: usage of two headphones and balanced
phones operation. Therefore plugging in phones into output
PH 1/2 brings up a warning that this output is coupled to the
rear outputs, and should only be used if required.
PH 1/2 stays deactivated until Dual Phones mode has been
activated via SETUP.
Overload / Short detected
An internal overload can be caused by too high output levels
and too low load impedance. A short circuit in the TRS plug
will also trigger the overload detection. In such a case the
relay will permanently disconnect the phone from the Extreme
Power output stage. If the phones plug is removed and
plugged in again after one second, the unit will activate the
phones output again.
This scheme has been introduced to force the user to check
cabling and connection. For example, a TRS plug not fully
inserted might stay unnoticed but can cause a short circuit.
DC detected
DC detection is crucial to prevent the sensitive phones drivers
from being destroyed by inaudible currents flowing through
them. As the ADI-2 Pro is fully DC coupled from DAC to
phones output, a digital full scale signal with 0 Hz would produce up to 15V DC at the outputs, and destroy any attached
phone immediately. In case the power output stage fails the
same could happen. Therefore the phones outputs will switch
off with 1.8V DC detected.
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Power Fail
In case the operating voltage drops below 9.3V the internal
power supply of the analog I/Os is switched off (overcurrent
protection). However the digital part will operate even with only
5V. Therefore connecting a wrong power supply could pretend
a fully working unit - which just doesn't receive or emit any
audio. This warning screen informs about the undervoltage
problem.
Internal Error
During power-on the unit performs a self-test. When this test
fails USB will be deactivated, with record and playback no
longer functional. In such a case please contact your local
RME distributor.
The ADI-2 Pro also shows certain Info Messages during normal operation, to explain the current state and to point out possible problems.
In AD/DA and DAC mode, a Non-Audio Channel Status causes the DA section to be muted. An
info message Non-Audio signal at SPDIF input gives a hint why there is currently no analog audio at the outputs present.
In USB mode, an Emphasis Channel Status brings up the info message Emphasis detected at
SPDIF input. This reminds the user that the Emphasis indication gets lost when recording
through a computer.
On any change of the Basic Operation mode, the new mode is shown for 2 seconds in the display (Preamp mode active, AD/DA mode active, USB mode active, Dig Through mode active).
When plugging in two connectors into the front with Balanced Phones mode Auto active, Bal-
anced Phones mode active will be shown briefly.
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17. Modes
17.1 Auto
The ADI-2 Pro is an AD/DA converter, USB audio interface, USB DAC, analog headphone amp,
format converter and digital monitoring device, with extended flexibility and versatility, equipped
with 5 input sources and 6 output paths. Usually that means an overflowing menu structure and
endless searches in the menus to get it working in even simple applications.
To prevent such frustrating situations the ADI-2 Pro includes an (and ships with activated) Auto
setup mode. When SETUP – Options – Device Mode /DSD – Basic Mode is set to Auto, the unit
will turn into different operating modes depending on the connected cables:
Preamp: Analog in to Analog out. This mode is active when no digital input signal and no
USB is detected.
AD/DA: Converter Mode, analog in to all digital outs, digital in to all analog outs. This mode
becomes active as soon as a digital signal is attached. This signal will also become the signal source. In case more than one digital signal is found the user has to manually select the
source to be monitored. The SRC is active as default and assigned to SPDIF. Clock mode is
slave to AES in, but will adapt to the detected source (Auto).
USB: or interface mode, or USB DAC. If USB is detected all inputs are routed to USB, all
outputs are fed from USB. USB has priority over the converter mode. In Setup the unit can
be configured as 2-channel or 6/8-channel device.
There are two more modes available, but not via Auto, only by manually activating them. Digital
Through Monitor (automatic clock and source selection of the digital inputs plus internal D to D
routing) allows to insert and monitor AES, SPDIF and ADAT signals. In case of ADAT only
channels 1/2 are monitored, but all 8 channels are passed through. DAC combines clock and
input source selection for easiest operation in home and HiFi usage.
In all modes (except DAC) Phones Out 3/4 is freely configurable, any possible source can be
monitored independently. This includes USB: when selecting one of the above modes manually
all inputs are still transferred to USB, and playback is possible when manually selecting USB as
signal source on analog output 3/4.
The unit remembers all settings, and loads these automatically when it is switched on again.
The following pages show block diagrams and include more details of these modes.
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17.2 Preamp
Preamp: Analog in to Analog out (internal digital routing).
This mode can be activated manually by selecting Basic Mode – Preamp. The device enters
Preamp mode automatically when Basic Mode is set to Auto and no digital input signal and no
USB is detected.
Analog input signals are automatically routed to the analog outputs. Use an analog source to
hear it via the rear analog outputs or the headphone outputs, amplified, EQ’d, processed, level
shifted and impedance or unbalanced/balanced converted.
Default sample rate in this mode is 192 kHz, which represents the optimum in fully available
DSP power and superior sonic transparency. Changing the sample rate manually is possible and
the new value will be remembered.
The below diagram shows that the analog input is sent to all digital outputs simultaneously. Any
DSP setting (EQ, phase etc.) of the analog input will affect all outputs. Furthermore the processed input signal is then processed independent for both analog outputs once more with the
respective output settings.
Note: To simplify the diagram and to maintain overview USB record is not shown. In all modes
all input signals are sent over USB to the host. In multi-channel mode three separate stereo
pairs, in stereo mode only the stereo analog input.
USB Playback channels 3/4 can only be monitored in multi-channel mode. In stereo mode selecting USB 3/4 plays channels 1/2.
The option Digital Out Source - Main Out (SETUP – Options – Device Mode / DSD) sends the
processed signal Main Out 1/2 to the digital outputs AES, SPDIF and ADAT, for example to
connect active monitors having digital inputs.
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17.3 AD/DA Converter
AD/DA: Converter Mode, analog in to all digital outs, digital in to all analog outs.
This mode can be activated manually by selecting Basic Mode – AD/DA. The device enters
AD/DA mode automatically when Basic Mode is set to Auto and a digital input signal is detected.
The detected digital input signal will also become the signal source. In case more than one digi-
tal signal is found the user has to manually select the source to be monitored (I/O – Output
Channel - Settings – Source). The SRC is active by default, assigned to SPDIF, but switched off
automatically if a DoP signal (DSD) is detected. Clock mode is slave to AES in, but will adapt to
the detected source (Auto).
Note: To simplify the diagram and to maintain overview USB record is not shown. In all modes
all input signals are sent over USB to the host. In multi-channel mode three separate stereo
pairs, in stereo mode only the stereo analog input.
USB Playback channels 3/4 can only be monitored in multi-channel mode. In stereo mode selecting USB 3/4 plays channels 1/2.
The option Digital Out Source - Main Out (SETUP – Options – Device Mode / DSD) sends the
processed signal Main Out 1/2 to the digital outputs AES, SPDIF and ADAT, for example to
connect active monitors having digital inputs.
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17.4 USB
USB: interface mode.
This mode can be activated manually by selecting Basic Mode – USB. The device enters USB
mode automatically when Basic Mode is set to Auto and a USB connection is detected. USB has
priority over AD/DA mode.
In USB mode all inputs are routed to USB, all outputs are fed from USB. In SETUP – Options –
Device Mode / DSD – CC-Mode the unit can be configured as 2-channel (Stereo) or 6/8-channel
(Multi-channel) device. Sample rates higher than 192 kHz and DSD128/256 are only available in
2-channel mode, which is why this mode has been chosen as default.
Class Compliant Stereo mode
In 2 channel / Stereo mode only the analog inputs are sent as USB record signal, and the stereo
USB playback signal is available simultaneously at all analog and digital outputs.
The two block diagrams show the small differences between both modes.
USB Playback channels 3/4 can only be monitored in multi-channel mode. In stereo mode selecting USB 3/4 plays channels 1/2.
The option Digital Out Source - Main Out (SETUP – Options – Device Mode / DSD) sends the
processed signal Main Out 1/2 to the digital outputs AES, SPDIF and ADAT, for example to
connect active monitors having digital inputs.
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Class Compliant Multi-channel mode
With USB connected all digital and analog inputs (6 channels) are routed to USB recording. In
the same way USB playback will feed all outputs separately (8 channels).
In 6/8 channel mode all I/Os can be used separately. Phones output 3/4 provides USB playback
of channels 1/2 when its Source is set to Auto (default).
The option Digital Out Source - Main Out (SETUP – Options – Device Mode / DSD) sends the
processed signal Main Out 1/2 also to the digital outputs AES, SPDIF and ADAT, for example to
connect active monitors having digital inputs (not shown). The playback channels 5/6 and 7/8
can not be used anymore.
Channel order in USB Operation
I/O Record Playback
1/2 Analog 1/2 Analog 1/2
3/4 AES Analog 3/4 (typically routed to Phones 3/4)
5/6 SPDIF (ADAT) AES
7/8 - SPDIF (ADAT)
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17.5 Digital Through Mode
This additional mode is a manual option only, it is not available via Basic Mode Auto. It has to be
activated manually by selecting Basic Mode – Dig Thru.
The purpose of the Digital Through Monitor is exactly what its name describes. A single digital
input signal is passed through the unit and can be monitored on the analog outputs at the same
time. Automatic clock and source selection of the digital inputs plus internal D to D routing allows
to insert and monitor AES, SPDIF or ADAT signals. With ADAT only channels 1/2 are monitored,
but all 8 channels are passed through.
The digital signal is not simply patched from input to output, but completely rebuilt. It gets totally
refreshed by SteadyClock FS, and even clock decoupled or up-/down sampled with the SRC
turned on. Without SRC the included 24 bit audio data is passed on bit-transparent.
As any digital input signal becomes available at all three digital outputs, Dig Thru mode also
offers digital format conversion as well as a distribution feature (one source to three destinations).
Note: To simplify the diagram and to maintain overview USB record is not shown. In all modes
all input signals are sent over USB to the host. In multi-channel mode three separate stereo
pairs, in stereo mode only the stereo analog input.
USB Playback channels 3/4 can only be monitored in multi-channel mode. In stereo mode selecting USB 3/4 plays channels 1/2.
The option Digital Out Source - Main Out is not available in this mode.
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17.6 DAC
This additional mode is a manual option and not available via Basic Mode Auto. It must be activated manually by choosing Basic Mode – DAC.
DAC simplifies operation and source selection when the ADI-2 Pro is used like a typical HiFi
DAC:
Simple 2 channel stereo operation
Easiest switching between the sources to be monitored, like USB and SPDIF
In this mode the source selection of Main Out 1/2 also defines the clock source of the unit.
Changing SPDIF/AES the ADI-2 Pro will automatically sync to SPDIF/AES input signal, thus
working in clock mode slave. When switching to USB the unit operates in clock mode master
(USB asynchronous).
The AD conversion always uses the last set sample rate of USB. Phones Out 3/4 always monitor
the same source signal as Main Out 1/2, but still have fully independent settings (Vol, EQ etc).
Class Compliant Stereo mode
In 2 channel / stereo mode only the analog inputs are sent as USB record signal, and the stereo
USB playback signal is available simultaneously at all analog and digital outputs.
The option Digital Out Source - Main Out (SETUP – Options – Device Mode / DSD) sends the
processed signal Main Out 1/2 also to the digital outputs AES, SPDIF and ADAT, for example to
connect active monitors having digital inputs.
Class Compliant Multi-channel mode
With USB connected, 3 stereo devices are active for recording and 4 stereo devices for playback. However, since the Phones output continues to follow the Main Out output, the analog
outputs can be used to listen to either channels 1/2 or 3/4, but not both at the same time.
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18. Balanced Phones Mode
In balanced operation, two identical power amplifiers are used to drive one side of the phones
each. Compared to normal, grounded operation, the voltage seen by the phone driver/speaker is
doubled. The power sent to it is even quadrupled.
With the comparatively low power required by headphones, and its already powerful Extreme
Power outputs, the balanced phones mode of the ADI-2 Pro has not been optimized for more
power, but more fidelity. See chapter 34.6 for all the details behind the extraordinary Advanced
Balanced mode implemented into the ADI-2 Pro.
In Advanced Balanced mode the ADI-2 Pro’s maximum output level rises to +13 dBu for HiPower Off and +28 dBu for Hi-Power On. The signal to noise ratio rises from 117 dB / 120 dBA
to 120 dB / 123 dBA. See chapter 34.18 for details about the available output power at the
phones outputs.
Balanced Phones mode requires headphones with separate cabling for left and right channel, 2
wires each, 4 wires in total.
Output PH 3/4 becomes the left channel
output, with former left channel as L+ and
former right channel as L−. PH 1/2 becomes
the right channel output, with former left
channel as R+ and former right channel as
R−. Gnd of both connectors stays uncon-
nected.
There exist different solutions on how to
connect balanced phones, but no standard
as such.
The use of a 4-pin XLR male connector on the phones side is quite popular. The diagram to the
right shows how an adapter cable can be used to connect the ADI-2 Pro phones outputs, using
two stereo TRS plugs and one female XLR connector.
See chapter 14.1.3 on how to activate Balanced Phones mode.
Notes: The Balanced Phones mode temporarily deactivates DSD Direct mode, in case Direct
DSD is set to On (menu then shows (ON) in brackets). The rear analog outputs 1/2 are muted
during balanced phones operation, because they are supplied with the same balanced phones
signal 1/2, which at the rear outputs represents an out-of-phase mono signal.
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19. DSD
19.1 General
DSD (Direct Stream Digital) is a stream with single bit resolution, but multiple times the sample
rate of the CD. DSD64 equals 64 times 44.1 kHz = 2.8 MHz, DSD128 5.6 MHz, DSD256 11.2
MHz. Versions with multiples of 48 kHz also exist, up to 12.2 MHz.
To transfer DSD data over SPDIF, AES or even USB, DSD over PCM (DoP) is the de-facto
standard. It uses only the lower 16 bit of a 24 bit word, the upper 8 bit are filled with a DoP
header signal to be able to detect it, and to lower the overall volume to prevent damage in case
of accidental playback as PCM data. Note that the data stays pure DSD and is NOT converted
to PCM.
The ADI-2 Pro supports DSD in various ways. When received via AES or SPDIF, the State
Overview screen will show DoP, and the DAC immediately turns from PCM to DSD mode. The
process is transparent to the user, playback will continue as usual. But there are no DSP functions possible on a bit stream. Therefore EQ, Crossfeed, Bass/Treble, Loudness and other audio functions provided by the DSP are deactivated. This is shown by adding brackets to the activated function, for example EQ Enable - (ON).
The DoP recognition also works on USB. A DSD playback via USB (State Overview shows
DSD) will be done at 176.4/192 kHz for DSD64, 352.8/384 kHz for DSD128, and 705.6/768 kHz
for DSD256. Under Windows WDM/WASAPI are currently limited to 384 kHz. The highest sample rates and DSD modes can only be used via ASIO. RME’s driver supports DSD over ASIO in
DoP format as well as ASIO native. ADI-2 Pro therefore is compatible to many programs, like
HQPlayer, and JRiver, but also to DSD recording software like Merging's Pyramix, Sound-It and
VinylStudio.
19.2 DSD Direct (Playback only)
To be able to digitally adjust the volume, DSD data must be converted to PCM. This is done
automatically within the DA converter chips. In DSD Direct mode there is no PCM conversion –
and consequently no volume control anymore. After having activated DSD Direct in the ADI-2
Pro’s menu (SETUP - Options - Device Mode / DSD - DSD Direct 1/2), channels 1/2 use this
mode when a DSD signal is received. The analog signal is then available at the rear outputs,
with a coarse volume control via the analog output reference level control. Outputs PH 1/2 are
deactivated. Outputs PH 3/4 stay in normal DSD mode and have the standard volume control
available.
The output level for digital full scale is 1.0 dB lower than with standard DSD mode. Therefore the
maximum analog output levels are 1.0 dB lower than the reference values. For a valid comparison between DSD and DSD Direct the volume of DSD should be set to -1.0 dB.
In mode DSD Direct balanced headphones can not be used. Therefore in Balanced Phones
mode DSD Direct mode is temporarily deactivated.
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19.3 DSD Playback
During a DSD playback all DSP functions of all channels are
temporarily disabled, even when transmitting PCM. This is
signalled in several menus by brackets around the (ON).
Analyzer and level meter show DSD signals in blue color,
the current mode is therefore easy to recognize.
The change between PCM and DSD causes a low volume
click noise. For optimized sound and fidelity, the ADI-2 Pro
does not use any lossy analog volume control, hence can't
suppress the DAC's click noise.
You may notice clicks and cracks at the title change quite often with DSD, even when the next
title has the exact same sample rate. This is caused by the 1-bit format, which, unlike PCM,
requires absolute silence and DC freedom at the beginning and end of a title, so that the transition as required by the 1-bit stream does not represent a random signal, which can sound like a
click or crack. Unfortunately, many freely available tracks are not 'clean' at the beginning and the
end. If these are played back one after the other by player software, the ADI-2 DAC's level meters show that the noise to be heard does not originate from the ADI-2 DAC, but is presented to
the DAC as a signal to be played. The level meters are in the digital domain before the DAC, so
proof of a faulty input signal is easy.
Note: A DSD playback via USB is only supported on playback channels 1/2. A DSD64 playback
in mode Multichannel via USB 3/4 does not activate the DSD mode.
With DSD playback, the analog output level is identical to a PCM playback. However, DSD limits
Volume control to +2.5 dB, while PCM allows up to +6 dB.
19.4 DSD Record
The ADI-2 Pro converts the analog input data not only to PCM, but optionally also to DSD. Via
I/O - Analog Input - AD Conversion the AD-converter can be switched from PCM (Default) to
DSD. Based on the current mode the DSD data are then sent to the outputs AES and SPDIF
(DoP), USB (DoP via ASIO or ASIO native), and the analog outputs 1/2 and 3/4 (re-converted by
the DAC).
All modes, routings, source options and block diagrams of chapter 17 are valid for the DSD operation as well, with both record and playback. The only difference: with DSD the circuit part
DSP is bypassed (no audio processing). In DSD mode all DSP functions of all channels are
temporarily disabled, even when transmitting PCM. This is signalled in several menus by brackets around the (ON).
A mixed operation is possible. During a DSD recording, also AD conversion, a PCM file can be
played back and monitored via DA. Even a playback of DSD via channels 1/2 and simultaneous
monitoring of AES or Analog In via output 3/4 is possible, no matter whether the sources are
PCM or DSD.
Limitations exist, but they are identical with PCM and DSD. For example the limit of 192 kHz on
the digital I/Os AES and SPDIF means that in mode AD/DA only DSD64 can be used, not
DSD128 and DSD256. That's not different to PCM, where the higher sample rates 384 and 768
kHz can't be used. And the common clock does not allow any combination of modes. For example a recording done at DSD64 (176.4 kHz) does not allow a simultaneous playback of 192 kHz;
again the same limitation applies in PCM mode.
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Software to record DSD audio:
Name
OS
URL
VinylStudio
Win/Mac
www.alpinesoft.co.uk
Sound-It!
Win/Mac
http://www.ssw.co.jp
Pyramix
Win
www.merging.com
AudioGate4
Win/Mac
www.korg.com
19.5 DSD Level Meter
While most DACs, even ones seen as 'Hi-End', leave the
user clueless during DSD operation, the ADI-2 Pro continues to show level as well as spectral content. To be able to
show the analog I/Os audio signals on level meters and
Analyzer the DSP performs an additional DSD to PCM conversion.
This additional conversion is not available for the digital
I/Os, shown in the Global Level Meter. Here DSD is shown
as it appears and sounds: with DoP as constant noise at
about -24 dBFS.
19.6 Beyond…
For the first time the ADI-2 Pro enables configurations, settings and applications that might give
answers to a lot of questions. Is there really a sonic difference between different DA-filters with
different impulse response properties? Just try it! Is there really a difference in sound between
DSD to PCM and DSD Direct? Just try it! Does an AD/DA converter chain sound different with
different sample rates? Just try it! Does the same converter chain sound different - and how
does it sound at all - if one uses DSD? Just try it!
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User's Guide
ADI-2 Pro FS R
Inputs and Outputs
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!
20. Analog Inputs
The ADI-2 Pro has two analog line inputs that can operate with levels up to +24 dBu. The electronic input stage uses a servo balanced design which handles unbalanced (TS jacks) and balanced signals (TRS / XLR) correctly, automatically adjusting the level reference.
When using unbalanced cables with the XLR inputs, pin 3 of the XLR jack should be connected to ground. Otherwise noise may occur, caused by the unconnected negative input of
the balanced input.
To use the inputs as unbalanced RCA: simply insert a standard TS male to RCA female adapter.
Now any RCA / Cinch cable can be easily used with the ADI-2 Pro.
One of the main issues when working with an AD-converter is to maintain the full dynamic range
within the best operating level. Therefore the ADI-2 Pro internally uses hi-quality electronic
switches, which allow for a perfect adaptation to the four most often used studio levels +4 dBu,
+13 dBu, +19 dBu or +24 dBu.
Additionally a digital Trim Gain of 0 to +6 dB, in steps of 0.5 dB, can be applied, to precisely
match the output level of external gear. The 6 dB of digital gain makes the whole range between
+13 dBu and +24 dBu available in steps of 0.5 dB. Due to the larger step of 9 dB between +4
dBu and +13 dBu, there is a small gap from +4 dBu to +7 dBu.
Trim Gain can also be used to increase the input sensitivity to -2 dBu for 0 dBFS. Note that digital gain reduces the basic signal to noise ratio of the ADI-2 Pro by the amount of the gain. In
real-world applications this will hardly be any problem, as the worst case SNR of -112 dBu (+6
dB Gain at +4 dBu) is very difficult to achieve from most analog sources.
Ref Vrms Digital Gain +6 / 0 dB Vrms
+24 dBu 12.28 +18 dBu to +24 dBu +18 = 6.15
+19 dBu 6.9 +13 dBu to +19 dBu +13 dbu 3.46 +7 dBu to +13 dBu +7 = 1.73
+4 dBu 1.23 - 2 dBu to +4 dBu - 2 = 0.62
The analog inputs also feature automatic overload protection. Activating the option Auto Ref
Lev(el) changes to the next higher reference level when overloads are detected.
21. Analog Outputs
21.1 General
The TS outputs, XLR outputs and the front outputs PH 1/2 are fed from the same DAC, hence
carry the same signal (channels 1/2). They all have individual driver stages with different output
levels, see next chapters for details.
All outputs feature mute components to suppress power on/off noise, even when power fails
instead of switching the unit off via the standby button.
SNR and THD values as well as frequency response are nearly identical on all analog outputs.
Both phones outputs feature mute relay, over-current detection, DC protection, plug detection
and DSP control, like auto assigning volume, volume ramp up, auto balanced mode, dual
phones mode, user interaction at overload detection, and low impedance level meter auto scaling.
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!
21.2 Line Out TRS 1/2
The ADI-2 Pro has two impedance balanced, short-circuit protected analog Line outputs that can
operate with levels up to +21.5 dBu (Ref Lev +19 dBu with Volume set to +2.5 dB). Using a stereo (TRS) jack the output uses two 100 Ohm resistors internally to achieve impedance balancing. Using an unbalanced mono (TS) jack ring is connected to ground automatically, the output
then operates unbalanced.
These outputs follow all Ref Lev settings, but stay at +19 dBu when +24 dBu is selected. For
Auto Ref Level see next chapter.
21.3 Line Out XLR 1/2
The ADI-2 Pro has two balanced analog outputs that can operate at levels up to +26.5 dBu (Ref
Lev +24 dBu with Volume set to +2.5 dB). The short circuit protected, low impedance line outputs 1/2 are available as XLR jacks on the back of the unit.
The XLR line outputs do not operate servo balanced! When connecting unbalanced
equipment, make sure pin 3 of the XLR output is not connected. A connection to ground
might cause a decreased THD (higher distortion) and increased power consumption!
To maintain an optimum level for devices connected to the analog outputs, the ADI-2 Pro internally uses hi-quality electronic switches, which allow for a perfect adaptation of all outputs to the
four most often used studio levels +4 dBu, +13 dBu, +19 dBu and +24 dBu.
The analog outputs 1/2 (and 3/4) also feature a mechanism to set the reference level automatically. Auto Ref Lev(el) maximizes the signal to noise ratio when using the Volume knob. It
changes to the next higher and lower reference level setting when the gain chosen via the Volume knob would suggest to set a better fitting value, with optimized signal to noise ratio.
Example: Ref Lev is set to +24 dBu, Volume is turned down by -21 dB. The effective signal to
noise ratio at the XLR output is now 120 dB minus 21 dB = 99 dB (RMS unweighted). While it is
unlikely that any noise will be audible, changing the Ref Level to +4 dBu would need only a Volume setting of -1 dB. The effective SNR then becomes 117 minus 1 = 116 dB. Such settings are
usually done manually by the user. Auto Ref Lev takes over this task, working in both directions
when turning Volume up or down.
Notes: This technique includes switching of hardware elements (like when doing it manually) and
is therefore not free of click noise. To prevent distortion the threshold for a Ref Level change
takes EQ gains into account.
21.4 PH Out 1/2
Channels 1/2 are also available on the front as 1/4" TRS (stereo) jack.
All outputs labelled 1/2 share the same Volume knob setting and the same hardware reference
level setting, with two small differences: as mentioned above the unbalanced TS output is
capped to +19 dBu, and the front output PH 1/2 has a 3 dB higher output level. So while it also is
capped at the +19 dBu Ref Lev setting, its real output level is +22 dBu.
This 3 dB gain turns +4 dBu into +7 dBu output level, +19 dBu into +22 dBu – only at the phones
jack PH 1/2. Reason: The output PH 1/2 becomes identical in level with output PH 3/4 (Hi-Power
Off = +7 dBu, Hi-Power On = +22 dBu). Identical levels for both phones outputs are required for
balanced phones operation, but also simplify usage and setup.
Please note that PH 3/4 is the main phones output of the ADI-2 Pro. PH 1/2 is designed as addition and extra functionality. Sharing volume/level settings with the rear outputs some limitations
arise. PH 1/2 is simply not totally independent, and should therefore only be used when really
needed. The ADI-2 Pro issues a warning message when inserting a plug into PH 1/2.
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If operation of the phones output 1/2 is desired, the Dual Phones mode has to be switched on.
The menu has the additional option to turn off the rear outputs as soon as PH 1/2 is plugged in.
Default is Mute On when plugged.
While these outputs are praised as ideal headphone outputs, eventually as well as technically
they also are ideal line outputs.
In case the output should operate as line output, an adapter TRS plug to RCA phono plugs,
or TRS plug to TS plugs is required.
The pin assignment follows international standards. The left channel is connected to the tip,
the right channel to the ring of the TRS
jack/plug.
RME has a history of phones outputs working perfectly as line outputs. The Extreme Power
phones outputs of the ADI-2 Pro continue this legacy by delivering exceptional performance with
widest compatibility and versatility. For example turning on Balanced Phones mode, the two TRS
jacks on the front perfectly serve as high-quality TRS balanced line outputs.
21.5 PH Out 3/4
The second 2-channel DAC in the ADI-2 Pro realizes a completely independent headphone output, PH 3/4. Channels 3/4 are available on the front as 1/4" TRS (stereo) jack. In any mode 3/4
can be set to any input, no matter which input is currently used by outputs 1/2. Using the SRC on
3/4’s input it can even run clock independent from output 1/2.
The Extreme Power driver stage is 100% identical to PH 1/2. To simplify usage of the main
phones output it has two output levels: Hi-Power off, equalling +7 dBu, and Hi-Power on, equalling +22 dBu. As explained above these match the settings +4 dBu and +19 dBu of PH 1/2.
Output PH 3/4 can also be used as unbalanced line output, see PH Out 1/2.
The menu includes an option to turn off the rear outputs as soon as a connector is plugged into
the PH 3/4 jack. Default is Mute On when plugged.
22. Digital Connections
22.1 AES
The ADI-2 Pro provides one XLR AES/EBU input and output each via the included breakout
cable when connected to the D-sub 9 pin socket on the back of the unit. Connection is accomplished using balanced cables with XLR plugs. Input and Output are transformer-balanced and
ground-free.
Input
The AES input is used when the unit is set to Auto and it is the only digital input signal. In case
USB is active and multi-channel mode has been set, the AES input signal is available as input
channels 3/4 when performing a USB recording. In 2-channel mode AES is only available for the
Phones Out 3/4 by manually selecting this input. Chapter 17 includes more details and block
diagrams for further explanation.
The AES input can be sample rate converted and clock de-coupled by activating the SRC on it.
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Output
As can be seen in the block diagrams of chapter 17, in most modes all digital outputs carry the
same signal. The ADI-2 Pro then operates like a splitter/distributor. The input signal is converted
to several digital formats at the same time, and can be used up to three times (AES, SPDIF
coaxial, SPDIF optical or ADAT).
In USB multi-channel mode the AES output becomes playback channels 5/6, see chapter 17.4.
The output signal coding of the ADI-2 Pro has been implemented according to AES3-1992
Amendment 4:
32 / 44.1 / 48 kHz, 88.2 / 96 kHz, 176.4 / 192 kHz depending on the current sample rate
Audio use
No Copyright, Copy permitted
Format Professional
Category General, Generation not indicated
2-Channel, No Emphasis
Aux Bits Audio use, 24 Bit
Origin: RME
Older AES/EBU (and SPDIF) devices and record media can contain Emphasis information. Audio signals with Emphasis have a high frequency boost, requiring high frequency attenuation
during playback. When using the ADI-2 Pro as audio interface to record SPDIF into an audio file,
the emphasis state is lost. See chapter 34.5 for details.
The option Digital Out Source - Main Out (SETUP – Options – Device Mode / DSD) sends the
processed signal Main Out 1/2 also to the digital outputs AES, SPDIF and ADAT, for example to
connect active monitors having digital inputs.
22.2 SPDIF
Input
Two SPDIF inputs are available, optical via TOSLINK and coaxial via the included breakout cable, but only one can be used at a time. The input is the white RCA connector.
The currently used SPDIF input can be chosen under Setup – Options – SPDIF / Remap Keys –
SPDIF In. Set to Auto the input with a valid input signal will be chosen automatically.
The optical input also understands ADAT format, up to 192 kHz, but only channels 1/2 are available.
To receive signals in AES/EBU format on the
coaxial input, an adapter cable can be used.
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.
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Output
With SPDIF 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 ADI-2 Pro as a splitter (distribution 1
on 2).
Under Setup – Options – SPDIF / Remap Keys – Optical Out the output format can be manually
changed from SPDIF to ADAT. Only two channels are sent via ADAT, the same that are still
available at the coaxial output.
The ADI-2 Pro’s SPDIF channel status has been implemented according to IEC60958:
32 / 44.1 / 48 kHz, 88.2 / 96 kHz, 176.4 / 192 kHz depending on the current sample rate
Audio use, Non-Audio
No Copyright, Copy Permitted
Format Consumer
Category General, Generation not indicated
2-channel, No Emphasis
Aux bits Audio Use
The option Digital Out Source - Main Out (SETUP – Options – SPDIF / Remap Keys) sends the
processed signal Main Out 1/2 also to the digital outputs AES, SPDIF and ADAT, for example to
connect active monitors having digital inputs.
Pin assignment of the 9-pin D-sub connector, breakout cable SPDIF / AES
Note: The digital breakout cable is identical to the one used in the DIGI96 and other HDSP series cards.
Pin
Name
Pin
Name
Pin
Name
1
GND
4
AES Out +
7
SPDIF In -
2
SPDIF Out +
5
AES In +
8
AES Out -
3
SPDIF In +
6
SPDIF Out -
9
AES In -
22.3 ADAT
The optical SPDIF input is fully compatible with all ADAT optical outputs. The internal receiver
detects the format and automatically switches between SPDIF and ADAT mode. RME's unsurpassed Bitclock PLL prevents clicks and drop outs even in extreme varipitch operation, and
guarantees a fast and low jitter lock to the digital input signal. A usual TOSLINK cable is sufficient for connection.
To not break the concept and to simplify operation, only channels 1/2 of the ADAT input signal
are taken. Even in multi-channel USB mode ADAT uses only the SPDIF input channels 5/6 – not
more. Still the information of all 8 ADAT channels is used in two cases: in SMUX (96 kHz) and
SMUX4 (192 kHz) operation, where channels 1/2 are built up from the information spread about
4 and 8 channels. And in Digital Through mode, where only channels 1/2 can be monitored, but
all 8 channels are passed through to the output if set to ADAT (see below).
ADAT input supports the unofficial signalling of double speed mode, 88.2 and 96 kHz. In slave
mode the clock will automatically jump into double speed mode than. Several RME interfaces
support such a signalling.
Under Setup – Options – SPDIF / Remap Keys – Optical Out the output format can be manually
changed from SPDIF to ADAT. Only two channels are sent via ADAT, the same that are still
available at the coaxial output.
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Installation and Operation – Windows
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23. Driver Installation
Note: When operated in CC mode Stereo the ADI-2 Pro is fully compatible to Windows 10 (1709
or newer). An installation of RME drivers is still recommended. They add ASIO (PCM, DSD DoP
and DSD Native) and 768 kHz WDM. They are also required for firmware updates and
DIGICheck. Additionally the Multi-channel mode is not fully working in Windows 10 (tested with
1803). Further information on operation without RME driver can be found in chapter 34.23.
RME is constantly improving their drivers. Please download the latest driver from the RME web-
site at http://rme.to/downloads, driver_madiface_win_09716.zip or newer. Unzip the downloaded file and start the driver installation by a double-click on rmeinstaller.exe. Follow the instructions of the installer. After installation connect computer and ADI-2 Pro. Windows detects
the new hardware as ADI-2 Pro and installs the drivers automatically.
After a reboot, the icon of the Settings dialog appears in the notification
area. Windows might hide it behind the triangle or upwards symbol.
Click on it to access it and to configure its appearance.
Chapter 34.3 explains how to find the ideal USB port.
Driver Updates do not require to remove the existing driver. Simply install the new driver over
the existing one.
Possible reasons why an ADI-2 Pro is not found automatically:
The ADI-2 Pro is not switched on
The USB port is not active in the system (check the Device Manager)
The USB cable is not, or not correctly inserted into the socket
Use the ADI-2 Pro State Overview screen to verify USB is detected and working (chapter
15.3)
De-installing the Driver
Usually a de-installation of the driver files is not necessary. Thanks to full Plug & Play support,
the driver files will not be loaded after the hardware has been removed.
Windows Plug & Play methods do not cover the registration of the ASIO driver. This entry can be
removed from the registry by a software de-installation request. This request can be found (like
all de-installation entries) in Control Panel, Programs and Features. Click on the entry 'RME
MADIface', then Uninstall.
To use Windows own Class Compliant drivers the RME driver has to be removed completely
(for example after a firmware update). In Device Manager select the ADI-2 Pro under Sound,
Video and Game Controllers, right click and choose Uninstall. In the next dialog make sure to
check 'Delete the driver software for this device'. Otherwise the driver will stay within the Windows installation and gets reinstalled automatically after the next reboot.
Firmware Update
Please see chapter 7. Under Windows the flash update tool requires the MADIface series driver
to be installed, see above.
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24. Configuring the ADI-2 Pro
24.1 Settings Dialog
Configuration of the ADI-2 Pro is usually done directly at the unit. For ASIO operation sample
rate and buffer size (latency) can be set via a dedicated settings dialog. The panel 'Settings' can
be opened by clicking on the fire symbol in the Task Bar's notification area
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 active, which means that any new settings might not be
applied immediately.
Buffer Size
The setting Buffer Size determines the latency between incoming and outgoing ASIO and WDM
data, as well as affecting system stability.
USB Diagnosis shows specific USB
transmission errors (CRC5/16, usually 0)
and general errors. If the unit detects a
record or playback error the number
shown will no longer be 0. An audio reset
is performed automatically. The counter
is reset on start of playback/record.
Sample Rate
Sets the currently used sample rate.
Offers a central and comfortable way of
configuring the sample rate of all WDM
devices to the same value, as since Vista
the audio software is no longer allowed
to set the sample rate. However, an
ASIO program can still set the sample
rate.
During record/playback the selection is
greyed out, so no change is possible.
The tab About includes information about the current driver and firmware version plus two more
options:
Lock Registry
Default: off. Checking this option brings up a dialog to enter a password. Changes in the Settings dialog are no longer written to the registry. As the settings are always loaded from the registry when starting the computer, this method provides an easy way to define an initial state for
the ADI-2 Pro.
Enable MMCSS for ASIO activates support with higher priority for the ASIO driver. Note: At this
time, activating this option seems to be useful only with the latest Cubase/Nuendo at higher
load. With other software this option can decrease performance. The change becomes active
after an ASIO reset. Therefore it is easy to quickly check which setting works better.
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24.2 Clock Modes - Synchronization
In the digital world, all devices must be either Master (clock source) or Slave (clock receiver).
Whenever several devices are linked within a system, there must always be a single master
clock.
A digital system can only have one master! If the ADI-2 Pro’s clock mode is set to 'Internal',
all other devices must be set to ‘Slave’.
To cope with some situations which may arise in studio practice, defining a sync reference is
essential. RME’s exclusive SyncCheck technology enables an easy to use check and display of
the current clock status. In the State Overview screen the column SYNC will show for all digital
inputs whether there is a valid signal (Lock, No Lock) for the optical input, or if there is a valid
and synchronous signal (Sync). See chapter 15.3.
The SRC (Sample Rate Converter) can be used to de-couple the clocking, allowing to use more
than one clock master in a digital setup. See chapter 8.6 for details.
Under WDM the ADI-2 Pro will (has to) set
the sample rate. Therefore the error shown to
the right can occur. An AES, SPDIF or ADAT
signal with a sample rate of 48 kHz is used as sync source, but Windows audio had been set to
44100 Hz before. The red color of the text label signals the error condition, and prompts the user
to set 48000 Hz manually as sample rate.
25. Operation and Usage
25.1 Playback
In the audio application being used, ADI-2 Pro must be selected as output device. It can often be
found in the Options, Preferences or Settings menus, as Playback Device, Audio Devices etc.
Increasing the number and/or size of audio buffers in the application (WDM) or the RME Settings dialog (ASIO) may prevent the audio signal from breaking up, but also increases latency
i.e. output is delayed.
Please note that currently Windows WDM is limited to 384 kHz. 768 kHz can only be used via
ASIO.
Note: Since Vista the audio application can no longer control the sample rate under WDM.
Therefore the driver of the ADI-2 Pro includes a way to set the sample rate globally for all WDM
devices, found within the Settings dialog. See chapter 24.1.
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25.2 DVD-Playback (AC-3/DTS)
AC-3 / DTS
When using popular DVD software players, their audio data stream can be sent to any AC3/DTS capable receiver via the ADI-2 Pro.
The sample rate must be set to 48 kHz in the ADI-2 Pro Settings dialog, or the software will
only playback the down-mixed analog signal via SPDIF.
In some cases an ADI-2 Pro output device has to be selected in >Control Panel / Sound / Playback< and be set as Default, or the software will not recognize it.
The DVD software's audio properties now show the options 'SPDIF Out' or similar. When selecting it, the software will transfer the non-decoded digital multi-channel data stream to the ADI-2
Pro’s AES and SPDIF output.
Note: This SPDIF signal sounds like chopped noise at highest level. Therefore the ADI-2 Pro will
automatically mute the analog outputs.
Multi-channel
DVD player software can also operate as software decoder, sending a DVD's multi-channel data
stream directly to the analog or digital outputs of the ADI-2 Pro. For this to work set CC-Mode
under SETUP - Options - Device Mode / DSD to Multi-channel, and the WDM playback device
’Loudspeaker’ of the ADI-2 Pro in >Control Panel/ Sound/ Playback< as ‘Standard’. Additionally
the loudspeaker setup, found under >Configuration<, has to be changed from Stereo to 5.1 Sur-
round.
The software's audio properties now list several multi-channel modes. If one of these is selected, the software sends the decoded analog multi-channel data to the ADI-2 Pro. On some software it is not necessary to select the Loudspeaker in the Sound panel.
25.3 Multi-client Operation
RME audio interfaces support multi-client operation. Several programs can be used at the same
time. The formats ASIO and WDM can even be used on the same playback channels simultaneously. As WDM uses a real-time sample rate conversion (ASIO does not), all active ASIO
software has to use the same sample rate.
Inputs can be used from an unlimited number of WDM and ASIO software at the same time, as
the driver simply sends the data to all applications simultaneously.
RME's sophisticated tool DIGICheck operates like an ASIO host, using a special technique to
access playback channels directly. Therefore DIGICheck is able to analyse and display playback
data from any software, no matter which format it uses.
25.4 Multi-interface Operation
The current driver supports up to three RME devices of the MADIface series. All units have to be
in sync, i.e. have to receive valid digital sync information. The ADI-2 Pro can therefore be used
simultaneously with a MADIface XT, MADIface USB, MADIface Pro, Fireface UFX+, or just another ADI-2 Pro. Under ASIO all devices are presented as one ASIO device with all available
channels as I/Os.
If one of the units is set to clock mode Master, all others have to be set to clock mode Slave,
and have to be synced from the master by feeding ADAT, AES or SPDIF. The clock modes of all
units have to be set up correctly in their Settings dialog.
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25.5 ASIO
Start the ASIO software and select ASIO MADIface USB as the audio I/O device or the audio
driver.
The sample rate is set by the ASIO application. The buffer size (latency) is set in the RME Settings dialog.
The number of available channels depends on the current Class Compliant mode: 2 channels
I/O when set to Stereo, 6 in / 8 out when set to Multi-channel. See chapter 14.1.3. Note: changing the CC-Mode requires to temporarily disconnect the ADI-2 Pro from the computer.
The ASIO driver supports sample rates up to 768 kHz in PCM format. DSD record/playback is
supported as DoP within ASIO as well as via ASIO native. ASIO Direct Monitoring (ADM) is not
supported.
26. DIGICheck Windows
The DIGICheck software is a unique utility developed for testing, measuring and analysing digital
audio streams. Although this Windows software is fairly self-explanatory, it still includes a comprehensive online help. DIGICheck 5.93 operates as multi-client ASIO host, therefore can be
used in parallel to any software, with both inputs and outputs (!). The following is a short summary of the currently available functions:
Level Meter. High precision 24-bit resolution, 2/8 channels. Application examples: Peak level
measurement, RMS level measurement, over-detection, phase correlation measurement,
dynamic range and signal-to-noise ratios, RMS to peak difference (loudness), long term peak
measurement, input check. Oversampling mode for levels higher than 0 dBFS. Supports visualization according to the K-System.
Spectral Analyser. World wide unique 10-, 20- or 30-band display in analog bandpass filter
technology. 192 kHz-capable!
Vector Audio Scope. World wide unique Goniometer showing the typical afterglow of a os-
cilloscope-tube. Includes Correlation meter and level meter.
Totalyser. Spectral Analyser, Level Meter and Vector Audio Scope in a single window.
Surround Audio Scope. Professional Surround Level Meter with extended correlation anal-
ysis, ITU weighting and ITU summing meter.
ITU1770/EBU R128 Meter. For standardized loudness measurements.
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.
Global Record. Long-term recording of all channels at lowest system load.
Completely multi-client. Open as many measurement windows as you like, on any chan-
nels and inputs or outputs!
To install DIGICheck, go to www.rme-audio.com, section Downloads / DIGICheck. Download
the latest version, unzip and run setup.exe. Follow the instructions prompted on the screen.
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Installation and Operation – Mac OS X
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27. General
The ADI-2 Pro is a UAC 2.0 Class Compliant device. Mac OS X has full UAC support built-in,
there is no driver installation required. Connect computer and ADI-2 Pro with a USB cable. Mac
OS X detects the new hardware as ADI-2 Pro (serial number).
The number of available channels depends on the current Class Compliant mode: 2 channels
I/O when set to Stereo, 6 in / 8 out when set to Multi-channel. See chapter 14.1.3. Note: changing the CC-Mode requires to temporarily disconnect the interface from the computer.
For firmware updates please see chapter 7.
27.1 Configuring the ADI-2 Pro
Configuration of the ADI-2 Pro is mostly done directly at the unit. When set to Clock Source Internal, Mac OS X will set the current sample rate.
Via Launchpad – Other –
Audio MIDI Setup the ADI-2
Pro can be configured for the
system wide usage. The Audio
window includes a menu to
select the sample rate. In Stereo mode up to 768 kHz are
supported, in Multi-channel
mode up to 192 kHz. The two
modes can not be selected
here, but have to be chosen at
the unit while it is disconnected from the computer.
Use Configure Speakers to
freely configure the stereo or
multi-channel playback to any
available channels.
Applications that don't support card or channel selection will use the device chosen as Input and
Output in the System Preferences – Sound panel. This setting is also available in the Audio
MIDI Setup via the gear symbol at the bottom of the window.
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27.2 Clock Modes - Synchronization
In the digital world, all devices must be either Master (clock source) or Slave (clock receiver).
Whenever several devices are linked within a system, there must always be a single master
clock.
A digital system can only have one master! If the ADI-2 Pro’s clock mode is set to 'Internal',
all other devices must be set to ‘Slave’.
To cope with some situations which may arise in studio practice, defining a sync reference is
essential. RME’s exclusive SyncCheck technology enables an easy to use check and display of
the current clock status. In the State Overview screen the column SYNC will show for all digital
inputs whether there is a valid signal (Lock, No Lock) for the optical input, or if there is a valid
and synchronous signal (Sync). See chapter 15.3.
The SRC (Sample Rate Converter) can be used to de-couple the clocking, allowing to use more
than one clock master in a digital setup. See chapter 8.6 for details.
27.3 Multi-interface Operation
OS X supports the usage of more than one audio device within an audio software. This is done
via the Core Audio function Aggregate Devices, which allows to combine several devices into
one. All units have to be in sync, i.e. have to receive valid sync information via a digital input
signal, then all channels can be used at once.
If one of the devices is set to clock mode Master, all others have to be set to clock mode Slave,
and have to be synced from the master by feeding AES, SPDIF, Word or ADAT. The clock
modes of all units have to be set up correctly in their Settings dialog.
28. DIGICheck Mac
The DIGICheck software is a unique utility developed for testing, measuring and analysing digital
audio streams. Although this software is fairly self-explanatory, it still includes a comprehensive
online help. DIGICheck 0.73 operates in parallel to any software, showing all input data. The
following is a short summary of the currently available functions:
Level Meter. High precision 24-bit resolution, 2/8 channels. Application examples: Peak level
measurement, RMS level measurement, over detection, phase correlation measurement,
dynamic range and signal-to-noise ratios, RMS to peak difference (loudness), long term peak
measurement, input check. Oversampling mode for levels higher than 0 dBFS. Supports visualization according to the K-System.
Spectral Analyser. World wide unique 10-, 20- or 30-band display in analog bandpass filter
technology. 192 kHz-capable!
Vector Audio Scope. World wide unique Goniometer showing the typical afterglow of a os-
cilloscope-tube. Includes Correlation meter and level meter.
Totalyser. Spectral Analyser, Level Meter and Vector Audio Scope in a single window.
Surround Audio Scope. Professional Surround Level Meter with extended correlation analy-
sis, ITU weighting and ITU summing meter.
ITU1770/EBU R128 Meter. For standardized loudness measurements.
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.
Completely multi-client. Open as many measurement windows as you like, on any chan-
nels and inputs or outputs!
To install DIGICheck, go to www.rme-audio.com, section Downloads / DIGICheck. Download
the latest version, unzip and run the installer. Follow the instructions prompted on the screen.
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Installation and Operation – iOS
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29. General
The ADI-2 Pro operates in Class Compliant mode (UAC 2.0), a standard that is natively supported by operating systems like iOS, Mac OS X, Linux and Windows 10 (since 1709). No proprietary drivers are required, the device will be directly recognized.
The ADI-2 Pro provides iOS devices with the professional analog I/O connections they lack.
Professional balanced and unbalanced line inputs and outputs, two Extreme Power headphone
outputs that excel with both high and low impedance headphones, extensive gain and level adjustments, AES, SPDIF and ADAT I/O connectivity, PCM record / playback at up to 768 kHz,
and DSD record/playback at up to 11.2 MHz (DSD256). An app like Neutron can actually play
files with up to 768 kHz sample rate and DSD 256 natively via the ADI-2 Pro in Basic Mode Stereo.
The ADI-2 Pro does not supply power to the iPad/iPhone. The latest Lightning to USB 3 Camera
adapter from Apple includes a Lightning socket to connect the standard Apple power supply,
allowing to charge the i-device while it operates in Class Compliant mode with the ADI-2 Pro.
30. System requirements
Any Apple iPad with at least iOS 6 or an iPhone with at least iOS 7
Apple iPad Camera Connection Kit or Lightning to USB adapter
iPad Pro: no active adapter, just a simple USB-B to USB-C cable or connector adapter
31. Setup
Connect the USB cable to the Camera Connection Kit/Lightning adapter/iPad Pro. Start the idevice and plug the Kit/adapter into the i-device’s socket. If everything works as expected, the
unit will be used for all audio I/O. Audio playback in iTunes will automatically be performed by
the ADI-2 Pro, using analog outputs 1/2 and 3/4.
Note: The i-device’s volume control is inactive during USB operation.
32. Supported Inputs and Outputs
When connected to an iPad, the analog input 1 works with mono apps, inputs 1 and 2 with stereo apps (both dual mono and stereo), and up to 6 inputs with multi-channel applications like
MultiTrack DAW and Music Studio. Garage Band supports all 6 inputs, but only two at a time.
Auria and Cubasis can record all 6 inputs simultaneously.
Playback will use analog outputs 1 and 2, or even more channels if the app supports such operation, like Auria and Cubasis, which allow to use all 8 output channels when the device has been
set to CC-mode Multi-channel.
In Class Compliant mode the default clock mode is Internal, and iOS typically sets 96 kHz or
higher. Any app can change / set the sample rate to a desired value, but not all apps include a
choice to select one. Setting the ADI-2 Pro (and with it the i-device) to slave mode by selecting
the AES or SPDIF input as clock source, the ADI-2 Pro will be synchronized to the external digital sample rate. With a wrong external sample rate heavy audio noise will occur (use of the SRC
will help in specific cases). Without an external signal the ADI-2 Pro changes to its internal clock,
with the sample rate set by iOS or the app in use.
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Technical Reference
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33. Technical Specifications
33.1 Analog Inputs
XLR
Input: XLR, servo-balanced
Input impedance balanced: 18 kOhm, unbalanced: 9 kOhm
Input sensitivity switchable +24 dBu, +19 dBu, +13 dBu, +4 dBu @ 0 dBFS
Digital Trim Gain range: 0 dB up to +6 dB
Signal to Noise ratio (SNR) @ +13/19/24 dBu: 120 dB RMS unweighted, 124 dBA
Signal to Noise ratio (SNR) @ +4 dBu: 119 dB RMS unweighted, 123 dBA
Frequency response @ 44.1 kHz, -0.1 dB: 5 Hz – 20.5 kHz
Frequency response @ 96 kHz, -0.5 dB: 3 Hz – 45.5 kHz
Frequency response @ 192 kHz, -1 dB: 2 Hz – 92.7 kHz
Frequency response @ 384 kHz, -1 dB: < 1 Hz – 124 kHz
Frequency response @ 768 kHz, -3 dB: < 1 Hz – 180 kHz
THD @ -1 dBFS: -116 dB, 0.00016 %
THD+N @ -1 dBFS: -114 dB, 0.0002 %
THD @ -10 dBFS: -125 dB, 0.000056 %
Channel separation: > 110 dB
TRS
As input XLR, but:
Input: 6.3 mm TRS jack, servo-balanced
33.2 Analog Outputs
1/2 XLR
Output level switchable +24 dBu, +19 dBu, +13 dBu, +4 dBu @ 0 dBFS
Signal to Noise ratio (SNR) @ +13/19/24 dBu: 120 dB RMS unweighted, 124 dBA
Signal to Noise ratio (SNR) @ +4 dBu: 117 dB RMS unweighted, 120 dBA
Frequency response @ 44.1 kHz, -0.1 dB: 0 Hz – 20.2 kHz
Frequency response @ 96 kHz, -0.5 dB: 0 Hz – 44.9 kHz
Frequency response @ 192 kHz, -1 dB: 0 Hz – 88 kHz
Frequency response @ 384 kHz, -1 dB: 0 Hz – 115 kHz
Frequency response @ 768 kHz, -3 dB: 0 Hz – 109 kHz
THD @ 0 dBFS: < -120 dB, 0.0001 %
THD+N @ 0 dBFS: -115 dB, 0.00018 %
Channel separation: > 110 dB
Output impedance: 200 Ohm
1/2 TRS (rear)
As output XLR, but:
Output: 6.3 mm TRS jack, impedance balanced
Maximum output level: +19 dBu
Signal to Noise ratio (SNR) @ +19 dBu: 120 dB RMS unweighted, 123 dBA
Signal to Noise ratio (SNR) @ +13 dBu: 118 dB RMS unweighted, 121 dBA
Signal to Noise ratio (SNR) @ +4 dBu: 115 dB RMS unweighted, 118 dBA
Output impedance: 100 Ohm unbalanced, 200 Ohm balanced
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Phones 1/2
As Output 1/2 TS, but:
Output: 6.3 mm TRS jack, unbalanced, stereo
Output impedance: 0.1 Ohm
Signal to Noise ratio (SNR) @ +22 dBu: 120 dB RMS unweighted, 123 dBA
Signal to Noise ratio (SNR) @ +7 dBu: 118 dB RMS unweighted, 121 dBA
Output level at 0 dBFS, Ref Level +19 dBu, load 100 Ohm or up: +22 dBu (10 V)
Output level at 0 dBFS, Ref Level +4 dBu, load 8 Ohm or up: +7 dBu (1.73 V)
THD @ +18 dBu, 32 Ohm load, 1.2 Watt: -110 dB, 0.0003 %
THD+N @ + 18 dBu, 32 Ohm load: -107 dB, 0.00045 %
THD @ +14 dBu, 16 Ohm load, 0.94 Watt: -110 dB, 0.0003 %
Max power @ 0.001% THD: 1.5 W per channel
See chapter 34.18 for detailed charts about the available output levels and output power.
Output Phones 3/4
As Output Phones 1/2, but:
Output levels at 0 dBFS: Hi-Power off +7 dBu, Hi-Power On +22 dBu
Balanced Phones mode
As before, but:
Output levels at 0 dBFS: Hi-Power off +13 dBu (3.46 V), Hi-Power On +28 dBu (19.5 V)
Output impedance: 0.2 Ohm
Signal to Noise ratio (SNR) @ +28 dBu: 123 dB RMS unweighted, 126 dBA
Signal to Noise ratio (SNR) @ +13 dBu: 121 dB RMS unweighted, 125 dBA
Output level at 0 dBFS, Hi-Power On, load 150 Ohm or up: +28 dBu (19.5 V)
Output level at 0 dBFS, Hi-Power Off, load 8 Ohm or up: +13 dBu (3.46 V)
Max power @ 0.001% THD: 2.9 W per channel
33.3 Digital Inputs
General
Lock Range: 28 kHz – 200 kHz
Jitter suppression: > 50 dB (2.4 kHz)
Accepts Consumer and Professional format
AES/EBU
1 x XLR, transformer-balanced, galvanically isolated, according to AES3-1992
Input sensitivity 1.0 Vpp
SPDIF coaxial
1 x RCA, transformer-balanced, according to IEC 60958
High-sensitivity input stage (< 0.3 Vpp)
AES/EBU compatible (AES3-1992)
SPDIF optical
1 x optical, according to IEC 60958
ADAT compatible
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33.4 Digital Outputs
AES/EBU
1 x XLR, transformer-balanced, galvanically isolated, according to AES3-1992
Output level 2.7 Vpp
Format Professional according to AES3-1992 Amendment 4
Single Wire mode, sample rate 44 kHz up to 200 kHz
SPDIF coaxial
1 x RCA, according to IEC 60958
Output level 0.75 Vpp
Format Consumer SPDIF according to IEC 60958
Single Wire mode, sample rate 44 kHz up to 200 kHz
SPDIF optical
1 x optical, according to IEC 60958
Format Consumer (SPDIF) according to IEC 60958
Sample rate 44 kHz up to 200 kHz
33.5 Digital
Clocks: Internal, AES In, SPDIF In, ADAT In
Jitter suppression of external clocks: > 50 dB (2.4 kHz)
Effective clock jitter influence on AD and DA conversion: near zero
PLL ensures zero dropout, even at more than 100 ns jitter
Additional Digital Bitclock PLL for trouble-free varispeed ADAT operation
Supported sample rates for external clocks: 32 kHz up to 200 kHz
Internally supported sample rates: 44.1 kHz up to 768 kHz
33.6 General
Included power supply: external switching PSU, 100 - 240 V AC, 2 A, 24 Watts
Standby power consumption: 50 mW
Idle power consumption: 10 Watts, Max. power consumption: 22 Watts
Idle current at 12 V: 850 mA (10 Watts)
Dimensions (WxHxD): 215 x 44 x 130 mm (8.5" x 1.73" x 5.1")
Weight: 1.0 kg ( 2.2 lbs)
Temperature range: +5° up to +50° Celsius (41° F up to 122°F)
Relative humidity: < 75%, non condensing
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33.7 Connector Pinouts
Pin assignment of the 9-pin D-sub connector, breakout cable SPDIF / AES
Note: The digital breakout cable is identical to the one used in HDSPe series cards.
Pin
Name
Pin
Name
Pin
Name
1
GND
4
AES Out +
7
SPDIF In -
2
SPDIF Out +
5
AES In +
8
AES Out -
3
SPDIF In +
6
SPDIF Out -
9
AES In -
TRS jack analog output
The 1/4" TRS jacks on the rear are wired according to international standards:
Tip = + (hot), Ring = - (cold), Sleeve = GND.
XLR Connectors
The XLR sockets of the analog I/Os are wired according to international standards:
1 = GND (Shield), 2 = + (hot), 3 = - (cold).
The servo-balanced input circuitry allows to use monaural TS jacks (unbalanced) with no loss in
level. This is the same as when using a TRS-jack with ring connected to ground.
The XLR outputs do not operate servo-balanced! When connecting unbalanced equipment,
make sure pin 3 of the XLR output is not connected. A connection to ground might cause
higher THD (distortion) and power consumption!
TRS Phones jack
The analog output channels feed two
Phones outputs via two independent
driver circuits.
In case these outputs should operate
as Line outputs, 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.
In Balanced Mode operation the TRS
outputs change from unbalanced stereo
to balanced mono operation. An adapter cable as shown to the right gives
balanced phones with 4-pin XLR connector access to the ADI-2 Pro outputs.
Using mini XLRs the pinout (signal to
pin number) is identical.
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34. Technical Background
34.1 Lock and SyncCheck
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 base. 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.
A digital system can have only one master! If the ADI-2 Pro uses its internal clock, all other
devices must be set to ‘Slave’ mode and be synced to the ADI-2 Pro’s clock.
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 SPDIF signal is applied to the ADI-2 Pro, the State Overview screen shows LOCK, i. e. a
valid input signal. 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. Both sample
rates have to be fully identical, not only in their frequency, but also in their phase relation. This
state is called Sync and also shown in the State Overview screen if present.
Example: The ADI-2 Pro is set to 44.1 kHz internal clock, and a CD player is connected to its
input. The State Overview screen will display the input signal and LOCK state. The CD player's
sample rate is generated internally as well, and thus slightly higher or lower than the ADI-2 Pro's
internal sample rate. Result: When reading out the data, there will frequently be read errors that
cause audible clicks and drop outs.
In order to display this problem the ADI-2 Pro includes SyncCheck. It checks all clocks used for
synchronicity. If they are not synchronous to each other (i. e. absolutely identical) the State
Overview screen will show LOCK. In case they are synchronous the screen shows sync.
In the example above the CD player can not be set to be clock slave, it will always use its internal clock (master). There are two solutions:
Set the Clock Source selection of the ADI-2 Pro to SPDIF. The ADI-2 Pro will now precisely
follow the input signal’s clock, the State Overview screen will show a stable sync for the
SPDIF input.
Set the SRC (sample rate converter) to SPDIF In. The SRC operates as clock decoupler, so
the ADI-2 Pro can stay on internal clock. In this case the State Overview screen will still show
LOCK for the SPDIF input as the two sample rates have no steady phase relation.
In practice, SyncCheck allows for a quick overview of the correct configuration of all digital devices. This way one of the most difficult and error-prone topics of the digital studio world finally
becomes easy to handle.
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34.2 Latency and Monitoring
Preamp Mode
In Preamp mode the analog input signal is converted to digital, processed by the DSP, then converted back to analog. An ADC and DAC have a specific delay caused by their internal oversampling and anti-alias filtering. This delay has been reduced significantly in the last years, and
is now so small that usually it is impossible to hear. The values of the AD and DA conversion are
shown in the table below. The total latency is around 11 samples higher due to the FPGA and
DSP exchanging data (22 samples at Quad Speed). At 44.1 kHz there are 23 samples delay,
equalling 0.5 ms. At 192 kHz total delay is 36 samples or 0.2 ms! That’s not far from a straight
wire.
Low Latency
The ADI-2 Pro uses the latest top AD- and DA-converters with special low latency filters, offering
exceptional signal to noise and distortion figures in combination with a super-fast conversion.
The exact delays caused by the AD- and DA-conversion of the ADI-2 Pro converter chips are:
Sample rate kHz
44.1
48
96
192
AD (5 x 1/fs) ms
0.11
0.10
0.05
0.026
DA Sharp (6 x 1/fs) ms
0.14
0.13
0.06
0.026
DA Slow (5 x 1/fs) ms
0.11
0.11
0.05
0.026
DA Low Disp (10 x 1/fs) ms
0.23
0.21
0.11
0.052
DA NOS (1 x 1/fs) ms
0.02
0.02
0.01
0.005
These values represent an important step in further reducing the latency in the computer-based
recording studio. The added latency - at least from AD- and DA-conversion - can simply be ignored.
USB Recording and Playback
Class Compliant mode is not different to other formats. The transfer of data in/out of the computer is done in form of buffers. The size of these buffers define the latency or delay that occurs
on the input and output path. The total delay from analog input to analog output, passing through
the computer and a DAW software, is called Round Trip Latency. On a typical computer, a RTL
of 5 to 10 ms can be expected.
Under Mac OS X Class Compliant works identical to standard audio. The latencies are therefore
the same, and – as usual – defined within the DAW software by setting the desired buffer size.
Under iOS Class Compliant is the only way to I/O audio at all, and usually provides similar performance for different interfaces.
Under Windows RME provides the MADIface series driver to use the ADI-2 Pro like any other
RME audio interface, with the same spectacular performance, although being a Class Compliant
device. Both WDM and ASIO are available. Latency under ASIO mainly depends on the buffer
size set in the driver’s Settings dialog.
Typical RTL values at 44.1 kHz sample rate for Windows ASIO:
Buffer size RTL
128 samples 7.3 ms
64 samples 4.4 ms
32 samples 2.9 ms
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34.3 USB Audio
An ADI-2 Pro can achieve a performance similar to a PCI or PCI Express based soundcard
when used with an optimal PC. Low CPU load and click-free operation even at 64 samples buffer size are indeed possible on current computers. However, using older computers a simple
stereo playback will begin to cause a CPU load of more than 30%.
A computer blocked for a short time – no matter if ASIO or WDM – will lose one or more data
packets. Such problems can only be solved by increasing the buffer size (and with this the latency).
The ADI-2 Pro features a unique data checking, detecting errors
during transmission via USB and displaying them in the Settings dialog. Additionally the ADI-2 Pro provides a special mechanism to continue recording and playback in case of drop-outs, and to correct the
sample position in real-time.
Like any audio interface the ADI-2 Pro should have a data transmission to the computer as undisturbed as possible. The easiest way to guarantee this is to connect it to its own bus, which
should be no big problem as most USB 2.0 interfaces are a double bus design. A check in the
Device Manager can be done as follows:
Connect the ADI-2 Pro to a USB port
Start the Device Manager, View set to Devices by Connection
Select ACPI x86-based PC, Microsoft ACPI-Compliant System, expand PCI Bus
This branch normally includes two entries of a USB2 Enhanced Host Controller. A USB Root
Hub can be seen, which then connects all USB devices, including the ADI-2 Pro. By reconnecting to a different port this view immediately shows at which of the two controllers the ADI-2 Pro
is connected. With multiple devices it can also be checked if they are connected to the same
controller.
Furthermore this information can be used to operate an external USB drive without disturbing
the ADI-2 Pro, by simply connecting the drive to the other controller. The information is also valid
for USB 3 ports.
Especially with notebooks it can happen that all internal devices and all the sockets/ports are
connected to the same controller, with the second controller not used at all. In that case all devices have to use the same bus and interfere with each other.
Experienced RME users remember the above text from the manuals of our other interfaces with
lots of channels. Compared to these the ADI-2 Pro has two advantages:
It can be switched into Stereo mode, using an isochronous audio stream of only two chan-
nels (pretty ridiculous)
In most of its use cases it is not required to work at lowest latency. Setting the ASIO buffers
to their highest value offers a much less critical and more stable recording and playback experience.
But even the 6/8 channel Multi-channel mode is, compared to the up to 70/70 channels that
RME supports via USB 2.0, still a minimal load.
One should not underestimate the effect on higher sample rates with PCM and DSD, though.
Those require to transfer data of multiples of the amount typical for a channel at 48 kHz:
Base 48 kHz 96 kHz 192 kHz/DSD64 384 kHz/DSD128 768 kHz/DSD256
Channels 2 4 8 16 32
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Now it should be clear why the above advice can be quite important even for an ADI-2 Pro. In
Multi-channel mode the numbers are even higher:
Base 48 kHz 96 kHz 192 kHz/DSD64 384 kHz/DSD128 768 kHz/DSD256
Channels 8 16 32 64 128
Although on the edge, 384 kHz would work. But 768 kHz - no way. As the ADI-2 Pro should work
under iOS as well, which has a system limit in transfer bandwidth, its USB transfer mode is limited to 192 kHz in Multi-channel mode. Fortunately that is no real limitation. The additional digital
I/Os activated then do not support higher sample rates than 192 kHz anyway.
But once again it must be noted: at 192 kHz the USB interface is challenged with data amounts
equalling 32 audio channels transfer, although only 8 are in use.
34.4 M/S Processing
The mid/side principle is a special positioning technique for microphones, which results in a mid
signal on one channel and a side signal on the other channel. This information can be transformed back into a stereo signal quite easily. The process sends the monaural mid channel to
left and right, the side channel too, but phase inverted (180°) to the right channel.
For a better understanding: the mid channel represents the function L+R, while the side channel
represents L-R.
During record the monitoring needs to be done in 'conventional' stereo. Therefore the ADI-2 Pro
also offers the functionality of a M/S-decoder. Activation is done in the Settings panel of the
Hardware I/Os via the option M/S-Proc.
The M/S-Processing automatically operates as M/S encoder or decoder, depending on the
source signal format. When processing a usual stereo signal, all monaural information will be
shifted into the left channel, all stereo information into the right channel. Thus the stereo signal is
M/S encoded. This yields some interesting insights into the mono/stereo contents of modern
music productions.
Additionally some very interesting methods of manipulating the stereo base and generating stereo effects come up, as it is then very easy to process the side channel with Low Cut, Expander,
Compressor or Delay. Looping this effect chain back into the DAW via AD-conversion often a
transformation back to stereo is missing. That's one of the reasons the ADI-2 Pro offers M/SProc also in its analog input channels.
The other application is to split a single analog channel signal to both analog inputs, then activate M/S-Processing and analyze the digital left channel only. This method performs mono
summing, which raises the ADI's incredible SNR by another 3 dB for even better measurement
analysis capabilities.
But the most popular application in music recording is the manipulation of the stereo width: a
change of the level of the side channel allows to manipulate the stereo width from mono to stereo up to extended (this feature requires an external mixer).
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34.5 Emphasis
In the early times of digital audio, with AD and DA converters of only 14 bit resolution, a technique was used that is also known from radio transmission: pre- and de-emphasis. The audio
signal is equalized to have treble boosted before the conversion. When played back an analog
treble filter (the term high cut seems a bit strong) is required. Overall the audible noise and distortion caused by the AD and DA conversion was hoped to be reduced this way.
Some older CDs were recorded with Emphasis, and indeed Emphasis is part of the Red Book
standard. Listening to them requires a filter on the playback side or their sound will seem too
bright. The playback of older digital recordings from tape might also require de-emphasis, and
even one of the first DAT recorders used Emphasis constantly.
Fortunately digital to analog converter chips have support for de-emphasis included. The ADI-2
Pro activates the DAC’s de-emphasis automatically when the current source is AES or SPDIF
and the Emphasis bit is set in the incoming Channel Status. The State Overview screen can be
used to track this state, a WARNING SPDIF EMPHASIS message will be shown.
Why warning? Because when using the ADI-2 Pro as audio interface to record SPDIF into an
audio file, the emphasis state is lost. Similarly there exists no mechanism to let the audio playback software control the emphasis state of the ADI-2 Pro’s DAC during playback of that rec-
orded file. An option De-emphasis On in the channel’s I/O menu allows for a manual activation in
such a case.
The ADI-2 Pro can also perform both pre- and de-emphasis
outside the DAC with just a single band of its Parametric
EQ. The emphasis filter is based on a simple first order RC
filter with time constants of 50 μs and 15 μs. The frequency
response curve looks like a low-Q treble boost with its +3 dB
point at 3183 Hz, and the upper shelving point at 10610 Hz.
At 20 kHz gain hits +9.49 dB.
For an inverted filter curve select band 5 with type shelf
active, set Q to 0.5, Frequency to 5.2 kHz and Gain to -9.5
dB. Similarly, a pre-emphasis is done with the same settings
but Gain to +9.5 dB.
As the measurement below shows these settings compensate the DAC's de-emphasis with 0.1
dB accuracy.
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34.6 Balanced Phones Mode
Headphones usually share one wire between left and right channel: the common ground, hence
operation is unbalanced. A different way to build a powerful output stage uses a balanced design. Both wires to the speaker are ‘phase’, there is no ground connection. This technique is
mostly used in car audio, as the operating voltage is limited to 12 Volt, and balanced operation,
here called bridging, delivers double the output voltage and four times the output power to the
speaker.
In balanced operation, two identical power amplifiers are connected to one side of the phone's
speaker each, and the input signal of one of the power amps is inverted in polarity (180°). When
one power amp sends out a positive voltage, the other one sends out the same as negative.
Thus the voltage seen by the speaker is double as high.
With the comparatively low power required by headphones, balanced phones mode still has
some interesting aspects:
Output level is doubled. With the ADI-2 Pro +22 dBu would rise to +28 dBu (a gain of +6 dB).
Now only few might have a headphone that requires that level, a scary 19.5 Volts in output
voltage. But when the phone is driven at the same volume and effective level as before, the
driver stages now operate at a 6 dB lower level. This can have a positive effect on THD and
linearity.
Output power is quadrupled. A driver stage that was designed for 1 Watt will deliver 4 Watt
when a second such stage is used with inverted input signal. That is a significant rise, which
would allow to either achieve output powers that even bassheads would fear, or allows for the
use of less powerful or simpler output stages than usual.
Complete ground-free operation will prevent EMI and floating voltage level problems in cer-
tain (seldom) cases.
One often mentioned aspect is questionable: having no common wire anymore the separation
between left and right is optimized. That is correct in theory, but has not effect in real-world, unless the headphone cable is in a state that would have to be called defective.
The below block diagram shows the standard way to turn normal phones output stages into balanced mode. As mentioned two stereo outputs are necessary, and one side of them has to be
fed by an inverted signal.
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This design, as common as it is, has several disadvantages:
an analog inverter stage has to be added to the signal path
the common mode situation of the signal at the phones is compromised by the difference
between + and – phase, caused by the analog inverter
multiple relays and laborious cabling with wires running back and forth from/to the PCB are
typical for such a design
The ADI-2 Pro deserves a different, better way to go balanced. The picture below shows RME’s
exclusive design as implemented in the ADI-2 Pro.
The ADI-2 Pro has two DACs and a powerful DSP. These ingredients allow for a much improved
version with several advantages:
The whole path from DAC to phones stays totally unchanged. Not a single relay or change in
cabling is necessary within the ADI-2 Pro.
The whole signal path from DAC to the headphone speaker is balanced (!)
Signal inversion happens fully transparent and lossless within the digital domain
As known from mono-summing with ADCs and DACs, the two channels of each DAC now
operate in that way. The signal to noise ratio rises by 3 dB.
Indeed the whole analog output chain is part of that mono-summing. Noise from the Gain
amp and the driver stages is identically reduced.
The same is true for THD, which is not only lowered by the smaller output voltages per amp,
but also by the common mode suppression that the phones' speaker realizes. Furthermore
small deviations in the DA output path’s hardware are further minimized by averaging.
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The Advanced Balanced mode of the ADI-2 Pro is as unique as brilliant. Balanced mode never
made as much sense as when implemented like done here!
In Advanced Balanced mode the ADI-2 Pro’s maximum output level rises to +13 dBu for HiPower Off and +28 dBu for Hi-Power On. The signal to noise ratio rises from 117 dB / 120 dBA
to 120 dB / 123 dBA. Or in other words: while the output level rises by 6 dB, the noise rises only
by 3. And therefore stays inaudible as in normal operation.
While the higher output voltage might be useful for some older exotic headphones, a four times
higher output power (around 5 Watts, per channel) does not make sense. Fortunately the current limit circuit in the ADI-2 Pro takes care of that and prevents the output power to rise higher
than 3 Watts, and below 2 Watts at impedances below 24 Ohms. See chapter 34.18 for details
about the available output power at the phones outputs.
The Advanced Balanced mode design does have one drawback though: it will work in DSD
mode, but not in Direct DSD mode, because PH 1/2 is switched off due to the missing volume
control.
34.7 SteadyClock
RME’s SteadyClock technology guarantees an excellent performance in all clock modes. Its
highly efficient jitter suppression refreshes and cleans up any clock signal.
Modern circuit designs like hi-speed digital synthesizer, digital PLL, 800 MHz sample rate and
analog filtering allow RME to realize a completely newly developed clock technology, right within
the FPGA at lowest costs. Despite its remarkable features, SteadyClock reacts quite fast compared to other techniques. It locks in fractions of a second to the input signal, follows even extreme varipitch changes with phase accuracy, and locks directly within a range of 28 kHz up to
200 kHz.
The further improved SteadyClock FS technology attenuates even lowest frequency jitter (>1
Hz), and provides an even higher jitter suppression at lowest self-jitter. Thanks to the efficient
jitter suppression, the AD- and DA-conversion always operates on highest sonic level, being
completely independent from the quality of the incoming clock signal.
Common interface jitter values in real
world applications are below 10 ns, a
very good value is less than 2 ns.
The screenshot shows an extremely
jittery SPDIF signal of about 50 ns jitter
(top graph, yellow). SteadyClock turns
this signal into a clock with less than 2
ns jitter (lower graph, blue). The signal
processed by SteadyClock is of course
not only used internally, but also used
to clock the digital output. Therefore
the refreshed and jitter-cleaned signal
can be used as reference clock without
hesitation.
The picture shows interface jitter which is measured directly at a word clock output, or on the
digital signal itself. The so called sampling jitter, usually in the range of a few picoseconds, is
also very low in the ADI-2 Pro FS. One way to show this is to send out a specially modulated
11.025 kHz sine from the analog output, then analyze the sampled result. Jitter products are
shown in the measurement as symmetrical sidebands, like narrow needles. SteadyClock does
not show obvious sidebands. Also remarkable are identical measurements when used with internal or external clock – a typical SteadyClock feature.
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34.8 ADI-2 Pro as Hardware I/O for Measurements
Audio measurement systems have been (and still are) quite expensive. Several years ago much
cheaper software based solutions started to replace the expensive references, whenever the
measurements did not require absolute accuracy. Although the software itself might be 100%
accurate, the hardware used as generator and analyzer is often just a consumer soundcard.
That limits signal to noise ratio, frequency response and distortion values to the ones of said
soundcard.
At RME not only the well-known references Audio Precision and Rohde&Schwarz are used, but
also simpler, sometimes even more flexible or unusual solutions. A long-time favourite is HpW
Works, a software analyzer and generator program that has been in use by RME developers for
more than 20 years. Most measurement diagrams shown in this manual were done with it. In
many cases the expensive systems are only used for verification.
One of the development ADI-2 Pro goals was to make it so good that it can serve as hardware
frontend for audio measurement software. Accepting a few limitations the hardware should be
capable to measure most of the audio interfaces, DACs, ADCs and analog equipment that is in
daily use by many. To achieve this the ADI-2 Pro must have superior tech specs.
The superior real-world specs listed and shown throughout this manual make the ADI-2 Pro one
of the best hardware frontends available. 120 dB dynamic range (RMS unweighted), zero hum,
support for different reference levels, super low-noise outputs, very low THD values, galvanically
isolated operation through battery power, special 384 kHz SPDIF mode via optical connection,
and very small level tolerances – the ADI shines on both your desk as well as the measurement
lab!
Tips for optimal measurement results
Improve the signal-to-noise ratio (and thus THD+N) at the analog input by mono operation
To do this, place the source on both inputs simultaneously using a split cable, and activate M/S
processing in the analog input. The left channel now has an SNR of 123 dB instead of 120 dB at
unchanged level reference.
Improve signal-to-noise ratio (and thus THD+N) at the analog output by mono operation
To do this, use a split cable to connect both outputs, which must output the same signal, to one
cable. This 'short circuit' uses internal resistors of the ADI to improve the SNR by 2.5 dB.
THD /THD+N Measurement
Measurements of the THD require an active notch filter in front of the analog input. If this is not
available, it is often sufficient to set the input level not higher than -10 dBFS, as the ADC will
then have a THD below -130 dB. For THD+N, however, the noise component increases and
limits measurements to approximately -108 dB.
Maximum resolution at the analog input
Whenever possible, the analog input should be set to the lowest ref level. While the SNR hardly
changes from +24 dBu to +4 dBu (120 to 119 dB), the noise level changes with the Ref Level.
This means: at +24 dBu the noise floor is -96 dBu, at +4 dBu it is -115 dBu. So for low-level signals, 19 dB lower signals can be detected.
Measuring Frequency Response
As most current devices support sample rates up to 192 kHz, a hardware frontend must be capable of 384 kHz, or it cannot fully measure the frequency response of a 192 kHz device. But
even then the result might be surprising. At 384 kHz the DAC used in the ADI-2 Pro uses a fixed
slow filter which causes an early drop at higher frequencies than when measuring at 192 kHz
sample rate with Sharp filter selected. Being keen to measure deviations in the range of ± 0.1 dB
with stellar accuracy that doesn't help much.
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Therefore RME added a digital compensation filter to the DA path of the ADI-2 Pro. This filter is
fine-tuned to improve linearity of the frequency response when self-measuring from DA to AD
(loop) at 384 kHz, and is only active at 384 kHz sample rate.
The picture below shows the ADI-2 Pro in loopback mode, XLR Out to In, at 384 kHz sample
rate. The red curve is the DAC's original frequency response, starting at 10 kHz (everything below is straighter than a line), showing its early decline with -0.5 dB already at 70 kHz. The black
line shows the available frequency response at 192 kHz sample rate with Sharp filters selected.
It gives an idea which area a frontend should be able to measure accurately.
The blue curve shows the effect of RME's digital compensation filter at 384 kHz sample rate.
The frequency response is now near flat up to 90 kHz, with minimal decrease in the area of the
192 kHz sample. With this filter the ADI-2 Pro can easily measure any 192 kHz (and lower, of
course) operated device with high accuracy, at fractions of a dB.
Naturally the added digital filter compensation will cause distortion whenever a signal is applied
whose level exceeds the gain of the filter at the respective frequency. This is largely, but not
completely, absorbed by the 2.5 dB volume reserve of the ADI-2 Pro FS R. For example at 130
kHz, the maximum undistorted digital level is no longer 0 dBFS, but -1 dBFS. This technical
limitation has no meaning in real-world operation. This technical limitation is of no significance in
practice, as full scale measurements are not performed for such high frequencies.
Max DA Level for Sine Measurements at 384 kHz Sample Rate
0 Hz up to 110 kHz: 0 dBFS
120 kHz: -0,5 dBFS. 130 kHz: -1 dBFS. 150 kHz: -1.5 dBFS. 155 kHz: -1 dBFS.
Ab 160 kHz: -0.5 dBFS.
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34.9 Noise Levels in Hi-Speed Modes
The outstanding signal to noise ratio of the ADI-2 Pro AD-converters can be verified even without expensive test equipment, by using record level meters of various software. But when activating higher sample rates, the displayed noise level will rise from -120 dBFS to -114 dBFS at 96
kHz, and –92 dBFS at 192 kHz. This is not a failure. The software measures the noise of the
whole frequency range, at 96 kHz from 0 Hz to 48 kHz (RMS unweighted), at 192 kHz from 0 Hz
to 96 kHz.
When limiting the measurement range from 20 Hz to 20 kHz (so called audio bandpass) the
value would be -120 dB again. This can be verified with RME's DIGICheck. The function Bit
Statistic & Noise measures the noise floor as Limited Bandwidth, ignoring DC and ultrasound.
The reason for this behaviour is the noise shaping technology of the analog to digital converters.
They move all noise and distortion to the in-audible higher frequency range, above 40 kHz.
Therefore the noise is slightly increased in the ultrasound area. High-frequent noise has a high
energy. Add the quadrupled bandwidth, and a wideband measurement will show a significant
drop in SNR, while the human ear will notice absolutely no change in the audible noise floor.
As can be seen in the next picture, the noise floor stays at a remarkably low level even outside
the hearing range. At sample rates of up to 96 kHz the noise shaping happens completely outside of the transmission range.
As is common in professional Digital Audio Workstations, the level meters
of the ADI-2 Pro are band
limited to 40 kHz, so do not
show the excessive noise
levels of 768 kHz and
DSD, but everything within
the audio range and a bit
above.
Also noteworthy are the
improved noise shaping
filters of the ADC used in
the ADI-2 Pro, with adaptation to the higher sample
rates provided.
Indeed the rise in noise over frequency is much lower than in former converter chips, where for
example at 192 kHz sample rate the wideband noise measurement would not reach -92 dBFS,
but only -79 dBFS.
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34.10 AD Impulse Responses
On the AD side the ADI-2 Pro offers four filters: Short Delay Sharp, Short Delay Slow, Sharp and
Slow. Basically these behave and operate exactly the same way as the filters already described
for the DAC. SD Sharp and Sharp offer the most linear frequency response and highest suppression of mirroring (aliasing) at high frequency input signals. SD Slow and Slow try to combine
a high aliasing suppression with an optimal impulse response, but start to act early within the
higher audible range at standard sample rates. See measurements in chapter 34.13. Note: SD
Sharp/SD Slow each have the same frequency response as the shown Sharp/Slow.
In the picture to the right the impulse responses of
filters Short Delay Sharp (left) and Short Delay Slow
(right) are shown, at 44.1 kHz. The source signal is
an analog single sample impulse (see picture NOS in
chapter 34.11). This one includes frequencies higher
than half the sample rate that must be removed from
the aliasing filter of the ADC. Therefore it is not possible to digitize such a signal at 44.1 kHz sample rate
without added rounding or pre/post ringing.
SD Sharp and SD Slow are IIR filters (also called
minimum phase),which are not phase-linear and
mostly cause post ringing. The advantage of IIR is
the very low latency of a few samples only, which is
welcome in a studio monitoring situation.
In the picture to the right the impulse responses of
filters Sharp (left) and Slow (right) are shown, again
at 44.1 kHz. Sharp generates the expected, Slow a
near perfect impulse response.
Both are FIR filters that cause post and pre-ringing.
FIR in general has a higher latency, but is phaselinear over the whole frequency range. Slow shows a
lower amplitude - the filter's treble decrease in the
audible range, but only very low pre and post ringing.
SD Slow and Slow work best at 88.2/96 kHz, because the decrease in the high frequency area then
happens outside the audible range. At the same time
the impulse response is near perfect – caused by
both the filter as well as the doubled sample rate.
In Preamp mode the analog signal is AD-, then DA-converted. In factory default state this happens at 192 kHz sample rate. The same impulse is recorded and played back in much better
quality, as the quadrupled sample rate samples the 44.1 kHz single sample impulse at least four
times, and the filters operate at a much higher frequency as well. Therefore the pre- and postringing is four times 'quicker', and as such only a quarter in length. Still Slow and NOS offer the
most perfect impulse responses.
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34.11 DA Impulse Responses
Short delay Sharp Sharp
Short Delay Slow Slow
The screenshots above show the analog output signal of the DAC filters, stimulated by a digital
single sample impulse at 44.1 kHz sample rate. While Slow has the most perfect response, it
looses around 1.2 dB already at 15 kHz, see chapter 34.14. Both Short Delay are IIR filters, the
other two are FIR type. FIR is phase linear over the whole frequency range.
NOS (Non-Oversampling, SuperSlow)
The DAC includes another filter which is called Super Slow in its data sheet. The impulse response looks perfect, but checking the output signal with an Oscilloscope reveals steps that are
more typical for so called Non-OverSampling (NOS) devices, so we renamed it NOS within the
DAC filter menu. Note that there is no audible distortion, the steps equal high frequency harmonics that are mostly higher than 20 kHz. Please also note that Slow and NOS filters cause much
more aliasing into the audio band and out-of-band noise than Sharp filters.
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SD LD (Short Delay Low Dispersion)
In theory, a filter should have as little phase deviation as possible over the frequency range,
have as short a settling time as possible, an acceptable decay time, and provide the maximum
possible frequency range without deviation. The stopband attenuation should be high to prevent
aliasing. A latency as low as possible would extend the application possibilities to more than just
listening to music.
The filter with the bulky name Short Delay
Low Dispersion approaches this ideal quite
well. It does not have the early treble
attenuation of the slow filters (see chapter
34.14), has a shorter settling time than Slow,
an average decay time like Sharp, a phase
maximum of only 9° at late 18 kHz (basically
phase-linear in the audible range), and a
latency of only 10 samples, so that it is also
well suited for professional real-time
monitoring. This makes it much more than
just a successful compromise.
34.12 Frequency Response Measurements
Note: 44.1 kHz to 192 kHz measured with filter Sharp on AD and DA..
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34.13 AD Filter Curves
34.14 DA Filter Curves 44.1 kHz
Note: Sharp/SD Sharp and Slow/SD Slow are congruent. NOS shows a very early drop. SD Low
Dispersion is almost as linear as Sharp/SD Sharp.
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34.15 Loudness
34.16 Bass / Treble
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34.17 Distortion Measurements
The following measurements show spectral analyses of the AD and DA conversion of the ADI-2
Pro FS R, whereby the analog input and output stages are naturally included. While the noise
and modulation behavior is identical for all devices, the AD and DA chips used have tolerances
in the height and distribution of the harmonics. The distortion shown here is slightly different for
each device, and can be a few dB lower or higher.
Inputs XLR / TRS*
*Signal source: APx555B, High Precision Sine Generator
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Outputs XLR and TS*
* Measured with active notch filter via ADCs of ADI-2 Pro FS in mono mode (M/S)
Output Phones TRS**
** Measured with active notch filter via ADC of ADI-2 Pro FS. +18 dBu output level @ 32 Ohm
equal 1.2 W (per channel)
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34.18 Extreme Power Charts
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34.19 Phones Distortion Comparison
34.20 Impedance based Level Meters PH 1-4
The horizontal level meters in various screens of outputs 1 to 4 show the digital level fed to the
DAC. With loads higher than 32 Ohms the level meter's display matches the real analog output
level (0 dBFS = +22 dBu). But at 32 Ohms the ADI-2 Pro delivers only +19 dBu, at 16 Ohms +15
dBu to the phones outputs, because a reasonable current limiting circuit prevents a too high
output power at lower load impedances. When in Hi-Power mode, the highest undistorted value
at 16 Ohms is -7 dB on the level meter. Consequently it should either change its colour to red
then, or rescale the highest level to -7. Only then the user would be clearly informed which maximum level can be issued by the device undistorted. As far as we are aware such a feature is not
available in any such device. That will change right now.
As the current limiter of the Extreme Power output stages
operates in finer resolution, its reaction can be evaluated
dynamically and be used as simplified impedance detection.
Which can then be used to re-scale the level meters.
Exactly that happens fully automated within the ADI-2 Pro.
The peak values to the right serve as pre DAC information
and stay unchanged, but the meter's yellow and red areas
get shifted and extended to the left.
With revised volume settings, where the level meter stays below the red area, the user can now
be 100% sure that the ADI-2 Pro operates absolutely free of distortion, even in the most extreme
applications.
Notes on functionality: The re-scaling happens not before the highest undistorted level is exceeded. The re-scaling stays visible until the phones plug has been removed. An ongoing overload situation causes the activation of the Overload warning screen and a disconnection of the
phones output.
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34.21 Digital Volume Control
The ADI-2 Pro deliberately avoids an analog level adjustment by means of a potentiometer. Its
digital version surpasses an analog one in practically every conceivable point. Typical disadvantages of setting with potentiometers:
Synchronicity errors lead to panoramic shifts and significant volume deviations left / right, in
particular near the end points of the adjustment range.
In the middle setting range, there is an increased crosstalk and changes in the frequency
response. Changes in the frequency response also occur at the end regions of the adjustment path.
The setting range for optimum volume adjustment is often too small, or at the lower or upper
end of the potentiometer's adjustment range.
Non-reproducible settings (except 0 and 11).
Higher THD/THD+N. A point well known to measurement technicians. As soon as an analog
potentiometer is in the signal path, the unstable contact between wiper and resistive track
causes noise, which contains both THD (distortion) and N (noise), even in the stationary
state. Thus the -110 dB of a DAC quickly gets reduced to for example -80 or -70 dB.
Special volume ICs, which activate different resistance values by means of numerous electronic
switches, avoid some of the above mentioned points. Unfortunately, even the best of these ICs
do not achieve either THD or dynamics of the DACs used in the ADI-2 Pro, thus would affect its
analog output signal.
However, none of this is an issue with RME's digital volume control!
In fact an analog volume control has a (theoretical) advantage in only one point, namely the
maximum signal to noise ratio at a higher level reduction. In reality, current circuitry overturns
the theory, and the SNR at the output of such a device is no better than that of a digitally controlled one. This is even more true the better the DA converter works and the less noise it has just like the ADI-2 Pro, which provides the maximum noise ratio over a wide level range of 20
dB, thanks to its four reference levels realized in the analog domain.
The most often cited issue of a digital volume control is an alleged loss of resolution at higher
attenuation. An example: 117 dB dynamic roughly equals 19 bit resolution. A volume attenuation
of 48 dB (8 bit) leaves 11 bit of resolution. Such a simple, but important details omitting argumentation, usually ends with: the music must sound distorted in quieter parts, and the signal to
noise ratio is down to a useless 69 dB.
The former is simply wrong, the latter irrelevant in practice. Indeed there is a reduced signal to
noise ratio, but it doesn't matter, as the noise was not audible before (below the hearing threshold), and is still not audible after lowering the level. And the reduced SNR also applies to devices
with potentiometers, since the potentiometer is never placed at the output, but in the middle of
the circuit, followed by further electronics which also add some basic noise.
The quality of the ADI-2 Pro's digital volume control is best shown by measurements. Hard times
coming up for convinced supporters of the analog control, because here it is very clear that the
disadvantages of a digital volume adjustment, such as roughness and distortions at higher attenuation, simply do not exist - at least with RME.
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The following measurement shows a digital full-scale sine of 1 kHz, 16 bits without dither, which
is reduced in level by 40 dB. Also shown are a full-scale sine of 1 kHz with 24 bit, at 60 dB and
93.8 dB level attenuation, which is the lowest volume setting the ADI-2 Pro offers.
A high-resolution FFT like HpW Works makes it possible to disassemble the signal into individ-
ual frequencies, and to identify unwanted components down to a level of -190 dBFS. The measurement shows that the undithered 16 bit signal does not produce any distortion or other tones
above -170 dBFS. So at a volume setting of -40 dB the measurable THD is -130 dB. At 24 bit a
volume setting of -60 dB also achieves -130 dB without distortion. And at a volume setting of -
93.8 dB there are still -93 dB THD measurable.
These results clearly show that distortion products of the digital volume control are not drowned
by the DAC's noise, but are not generated at all. It works perfectly even with an undithered 16 bit
signal, no detectable distortion products are produced.
If the volume control is measured at the analog output, the demonstrable THD is reduced to
around -100 dB at a volume setting of -60 dB, by the self-noise of the DAC (SNR 117 dB RMS
unweighted). In the above measurement that would be seen as straight noise floor at -160
dBFS. The digital volume control of the ADI-2 Pro therefore works much more precisely and
cleaner than required for current top-level DACs.
In summary:
RME's digital volume control in 42 bit TotalMix technology avoids all the disadvantages of analog
level control via pots, is easy to use, offers reproducible settings, and the highest sound quality.
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34.22 Bit Test
A bit test is used to check the playback path for unwanted changes in the playback data. Playback software can cut bits, add dither, or change the level - without these changes becoming
noticed easily. A poorly programmed driver can manipulate bits, and a playback hardware could
be both badly designed and defective (hanging bits, swapped bits). Even such features as proper channel assignment, left/right synchronicity and polarity can be tested by a well-made bit test.
With a bit test, such errors can be detected and - more importantly - excluded.
How does it work?
Most bit tests take some time and are loud and unpleasant when playing through headphones or
speakers. RME uses a unique bit pattern, with defined levels and pauses. This consists of only
400 samples (<10 ms), and sounds like a dull, medium-loud click - harmless for ears and
equipment. The short, but efficient test sequence allows to check for the following changes and
errors:
Level changes, equalization, dynamic processing, polarity, channel swapping, sample offset,
hanging or twisted bits, dither, bit reduction.
The signal reaches the ADI-2 Pro via USB, AES or
SPDIF/ADAT. The unit has three continuously running
check circuits. If the test signal is detected correctly, the
device's display shows a message: Bit Test 16 bits, 24 bits
or 32 bits passed, depending on the detected signal. If the
transmission path is not bit-transparent or bit-accurate, the
signal only minimally changed, the message is not shown,
the bit test failed. There is no error notice.
RME provides several audio files as free download: 44.1, 96 and 192 kHz in 16 bit, 24 bit and 32
bit. These files in WAV format can be played easily on Windows, Mac OS X and Linux. For ease
of use (looping, players with fade in/out), the files contain the bit pattern several times. Runtime
is about 4 seconds.
Download:
http://www.rme-audio.de/download/bit_test_wavs.zip
The Zip archive contains:
441_16_adi2pro_bittest.wav 441_24_adi2pro_bittest.wav 441_32_adi2pro_bittest.wav
96_16_adi2pro_bittest.wav 96_24_adi2pro_bittest.wav 96_32_adi2pro_bittest.wav
192_16_adi2pro_bittest.wav 192_24_adi2pro_bittest.wav 192_32_adi2pro_bittest.wav
Theoretically, the use of the 32 bit file is sufficient. If the lower bits on the transmission path are
simply truncated, the corresponding message appears with the respectively recognized bit resolution, i.e. 24 or 16 bits.
Notes:
iOS, AES, SPDIF and ADAT are limited to 24 bit.
Some players in Mac OS X offer a Direct Mode, using 32 bit integer in non-mixable format.
The 32 bit test might still fail. At this time only HQPlayer 3.20 is known to pass.
SPDIF/ADAT and AES are checked behind clocking and SRC. Therefore the unit needs to be
synchronized correctly to the digital input signal, with the SRC being deactivated (default: SPDIF
In, active).
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34.23 Operation in the Hi-Fi Environment
The ADI-2 Pro is a gem not only with professional applications but also when using it at home
with the stereo system. While users working in the studio environment know everything about
reference levels and all kinds of connectors, others get confused by the missing RCA sockets
and have doubts that the unit can be used with Hi-Fi at all - and how to connect it then. This
chapter answers these questions
Can I switch the inputs by remote control?
The ADI-2 Pro FS R has five inputs which can be connected to up to five outputs. Therefore a
simple input selection is basically not possible. In addition, the different sources (AES, SPDIF,
USB, Analog) require a complex clock management, which in turn requires additional settings
when switching.
If the ADI-2 Pro is used like a conventional DAC, the Basic Mode DAC (manual option) helps in
combination with the CC-Mode Stereo (factory default). This limits the device to an operation
with stereo I/O, while retaining essential advantages such as separate volume and EQ settings
for Line and Phones. Here the currently used clock source is the one of the selected input.
In Basic Mode DAC I/O - Settings - Main Output 1/2 - AD/DA Source determines the source of
the monitored signal, also for Phones Out 3/4. Available sources are: AES, SPDIF c(oaxial),
Optical (SPDIF/ADAT), Analog (XLR/TRS) and USB 1/2. Switching is possible through this
menu directly on the device, and also via the remote control, since the sources are factory default for keys 1, 2, 5, 6 and 7 for the function Remap Keys.
Proceed as follows to switch to Basic Mode DAC:
Press the SETUP key on the device.
Turn the lower encoder 2 until Device Mode / DSD appears
Press Encoder 2 once so that Basic Mode appears highlighted
Turn encoder 2 clockwise until DAC appears
Press the key SETUP again to exit the menu
The input to be monitored can now be selected via the remote control:
Key Source Labeling
5 SPDIF c COAX
6 Optical OPT
7 USB USB
1 Analog 2 AES -
After pressing one of the keys, an info message appears. If the Basic Mode DAC is not active,
this is also indicated, since the input switchover is then not available.
Note: SETUP - Options - SPDIF / Remap Keys allows you to assign one of 52 commands/actions to each of the seven programmable keys on the remote control. Any assignment
is possible and can be stored as part of the device setup.
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How do I connect the device to my other devices that have only RCA?
By a simple adapter mono 6.35 mm to RCA (also called Phono and
Cinch). The adapters are plugged into the rear inputs and outputs done. Now the existing RCA cables can be used with the ADI-2 Pro. The
adapters can stay plugged in all time.
Cables with mono 6.35 mm on one side and RCA on the other also exist, and work perfectly.
The plug-in adapter solution has the advantage that the user can use his preferred RCA cables.
Will this cause a deterioration in sound?
No, for two reasons. The input of the ADI-2 Pro has an impedance of 9 kOhm, which is high
enough to not overstrain even older Hi-Fi units with higher impedance outputs (up to 1 kOhm).
That is even more true with newer units having outputs significantly below 1 kOhm. Additionally
the input of the ADI-2 Pro has been designed to have the exact same technical specifications,
no matter if a balanced (XLR) or unbalanced (mono 6.35 mm) signal is applied. RME's servobalanced input also provides an automatic level correction - so that even the reference levels are
identical.
With the outputs such an adapter causes no change at all - technical specifications and operation stay unchanged. The outputs of the ADI-2 Pro serve RCA inputs without any problem.
Do the advantages of the ADI-2 Pro's fully symmetrical design still exist when using unbalanced connections (RCA)?
Yes. Because the unbalanced input signal is internally converted to balanced directly after the
first input stage. On the output side RME uses a specially developed servo-balanced DAC filter,
which provides both paths of the balanced design to have full noise and distortion ratios. Therefore the technical specifications are reached even if the balanced XLR output is used unbalanced, disconnecting one pin - the required signal optimization is already done in the unit. At the
typical RCA output, the TS socket with plugged in adapter, an additional conversion balanced to
unbalanced is performed directly at the output. These efforts guarantee the supreme sound
quality of the ADI-2 Pro to be available in all operation and connection scenarios.
Which level setting is recommended?
As professional device the ADI-2 Pro offers input and output levels up to +24 dBu (12.24 V
RMS), which overwhelm standard Hi-Fi equipment. But that's no problem as a total of four reference levels are available. With Hi-Fi the setting +4 dBu (equals +1.78 dBV or 1.23 V RMS) is
recommended. This setting causes an ADI-2 Pro output level similar to that of many CD-players.
In case it is still too low simply change to +13 dBu (+10.8 dBV, 3.46 V RMS).
If the input level is still too low at a setting of +4 dBu, because the feeding unit's output level is
insufficient, I/O - Analog Input - Trim Gain will help to achieve an up to 6 dB higher level.
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Doesn't such a low level cause a significant increase in noise?
Usually yes - but not with the ADI-2 Pro. Switching the reference levels is done in the analog
domain, in hardware. The circuit has been optimized for near maximum signal to noise ratio
even at +4 dBu. Detailed values can be found in chapter 33.1 / 33.2. The input's signal to noise
ratio can even be verified by anyone, using the free tool DIGICheck, function Bit Statistics &
Noise (see chapter 26 / 28). Please note that for measuring the signal to noise ratio or dynamic
the inputs need to be shorted (0 Ohm).
The loss of only about 1 dB of noise ratio at +4 dBu is an outstanding achievement of engineering. However, in real world situations DIGICheck will offer a lot more revealing information. As
soon as something is connected to the inputs of the ADI-2 Pro one has to say goodbye to those
dream values. The basic noise and hum, especially from Hi-Fi units, is magnitudes higher …
Fixed Output Level – Lock Volume
The option Lock Volume in the I/O menu allows to lock the analog output level to a specific value, a change via the VOLUME knob is no longer possible. The Volume adjustment within the
menu is used to set the 'fixed' output level.
For example when the analog output should operate like a typical HiFi unit @ 2 V output level
(equals +8 dBu), set the hardware Ref Level to +13 dBu and Volume to -4.5 dB. More examples:
1 V (0 dBV, +2.2 dBu): Ref +4 dBu, Vol -2.5 dB
0.775 V (0 dBu, -2.2 dBV): Ref +4 dBu, Vol -4 dB
0.5 V (-3.8 dBu, -6 dBV): Ref + 4 dBu, Vol -9 dB
0.315 V (-10 dBV, -7.8 dBu): Ref +4 dBu, Vol -12 dB
Note that most analog reference levels (except 2V) usually offer a lot of headroom. Therefore
matching the levels to 0 dBFS might result in too low volume. Also note that the above table is
no longer valid when PEQ and Bass/Treble are in use. The digital level will be higher and might
even overload, which can be clearly seen on the output's level meter. Volume has to be further
reduced then.
How can I quickly change between USB playback and the digital input to monitor these?
The ADI-2 Pro actually addresses this case with its Auto mode – which fails when the USB cable
remains in the ADI-2 Pro, and the computer is still powered on. The switching of the source of
the Main Output 1/2 is blocked in USB mode, as it is used for playback. The solution is to manually change the Basic Mode from Auto to AD/DA. That's all. In factory configuration the ADI-2
Pro will recognize a digital signal at the coaxial or optical input, and immediately play it through
Mains Out 1/2.
The Basic Mode is located in the menu SETUP - Options - Device Mode / DSD. As the unit remembers the last menu selected, the entire operation is ideally done by pressing the SETUP key
and turning encoder 2 to toggle between USB (or Auto) and AD / DA.
The unit's state including the currently selected input can also be stored as Setup, and these
Setups then assigned to the function keys or the remote as quick access. This is how it works:
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First, the current state is stored as Setup 1 during USB playback:
Press the SETUP key, turn encoder 1 (menu Setups appears), turn encoder 2 until the field Set-
up Select shows the choice Store 1. Now press encoder 2 until the cursor has jumped down to
the lowest field and the setup has been saved (or alternatively enter a different name in-between
- but this can also be done later).
Now reconfigure the device for playback of the digital source - Basic Mode AD/DA - Clock
Source SPDIF etc. Next, this state is stored in the memory location 2. Operation carried out as
above, but this time with the selection Store 2.
In the same menu you can now load Setup 1 or Setup 2. A direct switching between these two
setups/states without having to go into the menu can be accomplished via Remap Function
Keys, in the menu SETUP - Options - SPDIF / Remap Keys. After activating the entry Remap
Keys (ON), the entries below are no longer grayed out. Setup 1 and Setup 2 can now be as-
signed to the VOL and I/O keys, for example.
After that pressing the VOL button loads Setup 1, that is, the USB mode, and pressing I/O loads
Setup 2, the 'Digital In' mode. The name of the setup appears briefly in a pop-up window, so
changing the name is worthwhile.
The original function of the respective function key, entering the menu, is still possible by pressing and holding the button for a brief moment (0.5 s).
The Basic Mode DAC offers an alternative, see first question.
My player does not switch the sample rate of the ADI to that of the played file during
playback
In CC-mode Stereo the ADI-2 Pro is fully compatible to Windows 10 (1709 or newer). An installation of the RME drivers is not necessary. The Windows CC drivers support automatic sample
rate switching via WASAPI Exclusive. DSD playback is also possible. On computers that do not
work optimally, there can also be a reduction in dropouts and clicks.
An installation of the RME drivers is nevertheless recommended, because they enable ASIO
(PCM, DSD DoP and DSD Native) and supplement 768 kHz WDM. The drivers are also required for a firmware update and DIGICheck. In addition, the multichannel mode in Windows 10
is incomplete (tested with 1803 and 1909).
To uninstall the RME drivers after a firmware update: In the Device Manager, select the ADI-2
Pro under Audio, Video and Game Controllers. Right click on its entry, then select Uninstall device. A dialog will appear. Important: Check the Delete driver software for this device box. Otherwise, the driver will remain in your Windows installation and will be automatically reinstalled
the next time you restart.
After a restart, use the Windows driver with all advantages of the latest firmware version of the
ADI-2 Pro. If the RME driver is still loaded, simply repeat the above procedure.
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User's Guide
ADI-2 Pro FS R
Miscellaneous
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35. Accessories
There are several items available for the ADI-2 Pro:
Part Number Description
NT-RME-2 (lockable) Power supply for ADI-2 Pro. Robust and light-weight switching power
supply, 100 V-240 V AC, 12 V 2 A DC. Lockable DC connector.
BO968 Digital breakout cable (9-pin D-sub to 2 x XLR and 2 x RCA)
USB Cable, 2m RME USB 2 cable, length 78” (2m)
Unirack Universal rackmount adapter (tub for two 9.5" devices)
MRC RME Multi Remote Control, infrared remote control
Optical cables for SPDIF and ADAT operation:
OK0100PRO Optical cable, TOSLINK, 1 m (3.3 ft)
OK0200PRO Optical cable, TOSLINK, 2 m (6.6 ft)
OK0300PRO Optical cable, TOSLINK, 3 m (9.9 ft)
36. Warranty
Each individual ADI-2 Pro undergoes comprehensive quality control and a complete test before
shipping. The usage of high grade components should guarantee a long and trouble-free operation of the unit.
If you suspect that your product is faulty, please contact your local retailer. Do not open the device by yourself as it may get damaged. It has been sealed with tamper-evident material, and
your warranty is void if those seals have been damaged.
Audio AG grants a limited manufacturer warranty of 6 months from the day of invoice showing
the date of sale. The length of the warranty period is different per country. Please contact your
local distributor for extended warranty information and service. Note that each country may have
regional specific warranty implications.
In any case 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.
No warranty service is provided when the product is not returned to the local distributor in the
region where the product had been originally shipped.
Audio AG does not accept claims for damages of any kind, especially consequential damage.
Liability is limited to the value of the ADI-2 Pro. The general terms of business drawn up by Audio AG apply at all times.
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37. Appendix
RME news, driver updates and further product information are available on RME’s website:
https://www.rme-audio.com
Worldwide distribution: Audio AG, Am Pfanderling 60, D-85778 Haimhausen, Tel.: (49) 08133 /
918170
Support via e-mail: support@rme-audio.com
List of international supporters: https://www.rme-audio.de/support.html
RME user forum: https://forum.rme-audio.de
Acknowledgements
The Bauer Binaural Crossfeed effect in the ADI-2 Pro was inspired by Boris Mikhaylov’s bs2b
implementation.
Trademarks
All trademarks, registered or otherwise, are the property of their respective owners. RME,
DIGICheck and Hammerfall are registered trademarks of RME Intelligent Audio Solutions.
SyncCheck, ZLM, DIGI96, SyncAlign, TMS, TotalMix, SteadyClock, ADI-2 Pro and Babyface Pro
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 7/8/10 are registered trademarks or trademarks of Microsoft Corp. Apple, iPad, iPhone and Mac OS are registered trademarks of Apple Inc. ASIO is a registered trademark of Steinberg Media Technologies
GmbH.
Copyright Matthias Carstens, 03/2021. Version 3.3
Current driver version Windows: 0.9734
Firmware: FPGA 250, DSP102, 03/2021
Although the contents of this User’s Guide have been thoroughly checked for errors, RME can not guar-
antee 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.
Note on Disposal
According to the guide line RL2012/19EU (WEEE – Directive on Waste
Electrical and Electronic Equipment), valid for all european countries,
this product has to be recycled at the end of its lifetime.
In case a disposal of electronic waste is not possible, the recycling can
also be done by Audio AG.
For this the device has to be sent free to the door to:
Audio AG
Am Pfanderling 60
D-85778 Haimhausen
Germany
Shipments not prepaid will be rejected and returned on the original sender's costs.
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38. Declaration of Conformity
CE
This device has been tested and found to comply with the limits of the European Council Directive on the approximation of the laws of the member states relating to electromagnetic compatibility according to RL2014/30/EU, and European Low Voltage Directive RL2014/35/EU.
FCC
This device complies with Part 15 of the FCC Rules. Operation is subject to the following two
conditions: (1) This device may not cause harmful interference, and (2) This device must accept
any interference received, including interference that may cause undesired operation.
Warning: Changes or modifications to this unit not expressly approved by the party
responsible for compliance could void the user's authority to operate the equipment.
Responsible Party in USA:
Synthax United States, 6600 NW 16th Street, Suite 10, Ft Lauderdale, FL 33313
T.:754.206.4220
Trade Name: RME, Model Number: ADI-2 Pro FS R
This equipment has been tested and found to comply with the limits for a Class B digital device, pursuant to Part 15 of
the FCC Rules. These limits are designed to provide reasonable protection against harmful interference in a residential
installation. This equipment generates, uses, and can radiate radio frequency energy and, if not installed and used in
accordance with the instructions, may cause harmful interference to radio communications. However, there is no guarantee that interference will not occur in a particular installation. If this equipment does cause harmful interference to
radio or television reception, which can be determined by turning the equipment off and on, the user is encouraged to
try to correct the interference by one or more of the following measures:
- Reorient or relocate the receiving antenna.
- Increase the separation between the equipment and receiver.
- Connect the equipment into an outlet on a circuit different from that to which the receiver is connected.
- Consult the dealer or an experienced radio/TV technician for help.
Caution: To comply with the limits of the Class B digital device, pursuant to Part 15 of the FCC
Rules, this device must be operated with computer equipment certified to comply with Class B
limits. All cables used to connect to the computer and peripherals must be shielded and grounded. Operation with non-certified computers or unshielded cables may cause interference to radio
or television reception.
RoHS
This product has been soldered lead-free and fulfils the requirements of the RoHS directive
RL2011/65/EU.
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