junger D02 Operation Manual

DIGITAL

DYNAMICS

PROCESSOR

D02

OPERATIONS MANUAL

rev. 3.1

Justus-von-Liebig-Straße 7, 12489 Berlin, Germany
Telefon: [phone] - 0 Fax: [phone] – 46
www.junger-audio.com

INTRODUCTION

The digital dynamics processor model d 02 is a professional
studio device that processes the dynamic range of digital , as
well as analog audio signals.

The unit comes with digital AES/EBU Interface and high
resolution 24 Bit A/D Converters, that allows dynamic range
processing (compressor, limiter, expander) in the digital and
analog domain.
The digital dynamics processor d 02 converts analog to digital
audio signals without the risk of clipping and overload. With the
combination of A/D-conversion (with headrom to avoid overload)
and the following digital processing of gain and limiter it is
possible to achieve the highest digital full scale signal without
clipping.
The increase in programme density and loudness level are
entirely free of the processing noises typical for dynamic range
prossesors, such as pumping, breathing or signal discolouration.
The unit is easy to operate and requires only a limited selection
of settings. All other parameters required for an inaudible
processing of the dynamic range are automatically controlled by
the programme signal and permanently optimized.

!

- fully digital processing device
audio data word length: 24 bit

- compressor, expander, limiter

- 4 presets (universal, pop music, speech, live)
for stereo or 2-channel-mode
complex, signal dependent control algorithms

- linear gain - 6 dB ... +15 dB, in 1 dB steps

- digital deemphasis filter

- multicoloured LED display
shows either input level, output level or gain change
with peak hold and digital full scale display

- digital audio interfaces
AES/EBU + S/PDIF + OPTICAL

- analog input, analog output
24 bit over-sampling ADC, 20 bit oversampling DAC
adjustable level, balanced

- redithering for 16 or 20 Bit digital output format

CONTENTS

0

1. The design of the device ..................................................

3

3
4
8

1.1. Basic functions ...........................................................

1.2. The Jünger Audio dynamics processor principle .......

1.3. A/D-conversion with digital full scale level ..................

2. Installation .........................................................................

9

9

9
10
11

2.1. Power supply ..............................................................

2.2. Connections ................................................................

2.3. Switches for configuration of the unit ..........................

2.4. Setting the Digital Reference Level .............................

3. Control and display elements ............................................

12

4. Functional description .......................................................

18

5. Application notes ..............................................................

23

23
23
24
25
26

26

5.1. Presets .......................................................................

5.2. Processing signals containing emphasis ....................

5.3. Working with headroom ..............................................

5.4. Influence of signal delay time ......................................

5.5. Selection of parameters to increase loudness ............

5.6. Redithering - Reduction of word length of digital output

signal ...........................................................................

6. Applications .......................................................................

28

7. Technical specification ......................................................

29

8. Warranty and service information .....................................

30

3

THE DESIGN OF THE
DEVICE

The d 02 digital dynamics processor can be used to process
both digital and analog audio signals. The device is primarily
designed for use with stereo signals.
Digital input signals can be connected in the AES/EBU standard
format, including SP/DIF and OPTICAL formats.
For the analog inputs high resolution 24 bit A/D converters are
used. The sample rate of the A/D-converter can be syncronised
to internal crystal clock generators or to external word clock
signals.
Input and output can be selected independently. The output
signals are available in parallel in all three digital formats so that,
depending on the active input, a format conversion can also be
achieved.

1.1.

Basic
Functions

1

In addition, an analog stereo signal output is available which
operates with 20-bit D/A converters and enables a rapid acoustic
monitoring.

The increase of signal density and loudness level of the digital
audio signals can be achieved by the interaction of two dynamic
range control processes. Firstly, by the compression achieved
by increasing low and medium signal levels and secondly, by
linear amplification combined with an inaudible limitation of
individual remaining peak levels by the limiter.

The outstanding quality of dynamic range processing is based on
the new Multi-loop dynamic range control principle developed
by Jünger Audio.

The term Multi-loop means that there are several interactively combined
control circuits as opposed to a control circuit with a spectrum split into
several bands with different frequencies (multi-band).

1. THE DESIGN OF THE DEVICE

4

1.2.

The
Jünger Audio
Dynamics
Processor
Principle

A change in the dynamic range of an audio signal is a non-linear
process. The gain of a dynamic range processor is not constant
as it is with the gain of a linear amplifier. The gain varies in time
depending on the input signal and depending on the specific
control algorithm of the dynamics processor. These variations in
the gain, which represent the real control process, should take
place without any bothersome side effects such as pumping,
signal distortion, sound colouration or noise modulation, which
means they should be inaudible.

The main problem here is to react to fast changes in the audio
signal (transients) without the control process being audible and
disturbing. The ability of a dynamic range processor to react to
rapid amplitude changes depends directly on its attack time.

Long attack times do not cause modulation distortions, but lead
to overshoots because the system is not fast enough to reduce
the gain. A short attack time minimizes the amplitude and time of
a possible overshoot, but a rapid gain change has audible side
effects such as " clicks" caused by modulation products.

Traditional compressor and limiter designs only have one control
circuit with corresponding attack and release times, which have
to be adjusted manually by the user. An optimal setting of all
parameters for dynamic range processing with as little
disturbance as possible must be determined by listening and
comparing.

traditional compressor
and limiter designs

A lot of experience and also a lot of time is necessary to get
sufficient results. These parameters , once found, are only the
right choice for a certain programme signal and must be changed
for other signals.
Dynamic range processors which split the audio frequency
spectrum into several bands, i.e. which have a multi-band
structure, have some advantages over traditional compressor
designs. The dynamic control parameters in each band are
independent of one another and can be set in such a way that a
broad program range can be processed well. Disruptive side
effects such as pumping and breathing can largely be avoided.
The disadvantage of this system lies in the problem of rebuilding
the output signal, which is the sum of all filters including those
where dynamic changes have taken place as part of the control
process.

multi-band structure

The output signal is always coloured and deviates from the input
signal in sound.

The dynamic range processor principle developed by Jünger
Audio makes it possible to realise dynamics processors
(compressor, limiter, expander) with very high audio quality,
without signal discolouration, pumping or breathing, without
distortion and modulation products - in short, with almost

1. THE DESIGN OF THE DEVICE

inaudible processing - and they are very easy to use.

The Jünger Audio dynamics processors work according to a
Multi-loop principle, operating with an interaction between several
frequency linear control circuits. The resulting attack and release
times of this system are variable and adapted to the evolution of
the input signal. This allows relatively long attack times during
steady-state signal conditions but also very short attack times
when there are impulsive input transients.

A A

f

delay

1

1

2

2

n

m

f

Multi - Band

Multi - Loo

p

The Multi-loop structure also permits a short time delay
between the control circuit and the gain changing element. The
gain control circuit has time to preview the signal and become
active before it reaches the output. This is particularly important
for the limiter, which provides a precisely leveled output signal
absolutely free of overshoots (clipping).

With a digital signal processor, a large number of parameters of
the audio signal are evaluated and there is a permanent,
automatic optimisation of the parameters of all control circuits.

Together with its attack and release times which determine the
dynamic qualities, the performance of a dynamic range
processor depends on the static compression characteristic.

The d 02 digital dynamics processor is a dynamic range
processor which, contrary to its conventional counterparts, is
effective for a wide dynamic range of input signals (50 dB).

multi-loop principle

fig. 1:

basic principles of
dynamic range
processors

delay time

compressor

5

1. THE DESIGN OF THE DEVICE

Figure 1 shows the basic principles of dynamic range processors.

The compression of the programme signal takes place evenly
over the entire range and not only at the upper end above a
certain threshold level. Dynamic structures of the input signal
(e.g. musical dynamic evolutions) are converted proportionally so
that even after compression the ratios are maintained, only
slightly condensed, leaving on the whole a transparent,
seemingly uncompressed sound impression.

Compression (reduction of the dynamic range of the input
signal to match the dynamic range of the storage or of the
transmission system) is partly achieved by increasing the level
of low level signals, the lowest of which might otherwise be
below the noise floor of the audio system. The lower the input
signal level the higher the additional gain applied to that input
signal by the compression processing will be.
Independent of the compression ratio , a maximum gain of the
compressor can be set, so that there can be no inadmissible
increase of background noises during signal pauses (e.g. live
atmos, air-conditioning, hum and noise).

compression gain

Below an adjustable threshold level an expander can be activated
which can lower the amout of noise signals.

expander

6

1. THE DESIGN OF THE DEVICE

fig.. 2:
static
characteristics:
compressor

2.0 : 1

1.6 : 1

1.3 : 1

1.1 : 1

off

static characteristics: compressor

compression gain: max. 10 dB

parameter: ratio

-50 -40 -30 -20 -10 0

-50

-40

-30

-20

-10

0

input level (dBFS

)

output level d02 (dBFS

)

off

5 dB

10 dB

15 dB

static characteristics: compressor

compression gain: max. 15 dB
parameter: compression gain

ratio: 1.6 : 1

-50 -40 -30 -20 -10 0

-50

-40

-30

-20

-10

0

input level (dBFS

)

output level d02 (dBFS

)

fig. 3:
static
characteristics:
compressor

7

1. THE DESIGN OF THE DEVICE

8

The usable dynamic range for digital recording is determined at
the top by the highest possible digital signal (full scale) and at the
bottom by the lowest possible digital resolution. This range
cannot be fully exploited when using a conventional analog­digital converter caused by the necessary headroom of 6 ... 10
dB to prevent over-level of the signal wich could otherwise occur.

1.3.

A/D­Conversion with
Digital Full Scale
Level

This headroom of 6 .. 10 dB reduces the signal to noise ratio by
the same amount even if a high quality A/D converter with 18 or
20 bit resolution is used.

It is therefore more important than noise-shaping or other
dither techniques to use primarily the maximum of available
digital dynamic range, because this improves most
effectively the signal to noise ratio.

The d 02 digital dynamics processor offers a unique combination
of a 24 bit A/D converter and a high quality digital limiter with
which a digital signal free of overload and with maximum digital
output level can be generated.

The A/D converter operates with normal headroom to avoid
overload. Then in the digital domain the level of the signal is
increased to the point where the limiter begins to control the level.
Any possible overload is corrected inaudible by the excellent audio
quality of the digital limiter.

9

INSTALLATION

The digital dynamics processor d02 is a device under the safety
category Schutzklasse 1 in keeping with the VDE 0804
standards and may only used with power supply installations
built according to regulations.
Check the voltage details printed at the rear panel are the same
as your local mains electricity supply.

All input and output connectors of the digital dynamics processor
d02 are arranged in functional groups on the rear panel.

LEFT RIGHTLEVEL

ANALOGOUTPUTANALOGINPUT

LEFT RIGHTLEVEL

OUT

AES/EBU S/PDIF

OPTICAL

OUT

IN

IN

model d02

230 V
50 Hz
200 mA

Nr

REMOTE

EXT SYNC

16

20
BIT

DIG OUT

24

CON

STATUS

PRO

2.1.

Power Supply

2.2.

Connections

2

POWER INPUT

IEC mains input connector 230 V, 50 Hz (UK: 240 V, 50 Hz; JAPAN: 100 V,
60 Hz; USA: 127 V, 60 Hz) with integrated fuse

REMOTE

for optional serial remote interface RS-232 input and output
connector: 15pin SUB-D, male

DIGITAL INPUTS AND OUTPUTS

AES/EBU

input and output for AES/EBU standard format
input: XLR female panel jack
1- open, 2-3 signal, balanced, max. 5 Vpp
output: XLR male panel jack
1- open, 2-3 signal, balanced, max. 5 Vpp

S/PDIF

digital format for semi-professional use
When a signal is present at the AES input at the same time it has
preference over SP/DIF
Input and output : RCA socket

OPTICAL
Optical interface for digital audio signals, (do not use input together with
SP/DIF input)
Input and output : TOSLINK

EXT SYNC

Word clock input for external synchronisation
Input and output: BNC, (TTL-level)

2. INSTALLATION

ANALOG INPUT

Analog input to 24 bit A/D-converter
Input electronically balanced, XLR connector female
adjustable level ( +12...+22 dBu for digital full scale)

ANALOG OUTPUT

Analog output from 24 bit D/A-converter
Output electronically balanced, XLR connector male

adjustable level ( +6...+22 dBu for digital full scale)

2.3.

Switches for
configuration of
the unit

Following switches in the mode field at rear panel are used for
configuration of the unit.

STATUS

Setting of sended channel-status-bits on digital

output by using of analogue input at any salmple rate.

Channel status bits

are defined in the AES/EBU data stream.

With the digital dynamics processor d02 it is possible to transmit
this information without changing or to set these information
defined.

(Sometimes it is helpful to change the channel status, f.i. if following units
don’t want to accept incoming signals.)

If using digital input of d02 unit is transparent for channel status
information. There is no changing or modification of it possible.
Channel status information at digital output is the same like
original digital input signal.

PRO selection of professional mode.

CON selection of consumer mode.

DIG OUT

Selection of dither mode for reduction of digital

output word length.

16 BIT Dither for reduction to 16 bit word length

20 BIT Dither for reduction to 20 bit word length

24 BIT Signal without dither (unreduced 24 bit word

length)

10

2. INSTALLATION

11

2.4.

Setting the Digital
Reference Level

The static characteristics of the processor d 02 are related to the
digital reference level.
This internal digital reference level is the maximum output level for
the limiter and the reference level for the static compressor
characteristics. The rotation point for the compressor
characteristics with zero gain is allways situated at the internal
digital reference level.

In order to adjust the digital reference level it is necessary to
change the operating mode of the unit as follows. Hold down
the display button continuously for a few seconds and the unit will
enter digital reference level adjustment mode. Pressing the INC or
DEC buttons on can change the digital reference level in the range
of 0 dBFS till -15 dBFS.

It is possible to store two different digital reference levels, one
for use when the analog signal input is selected and another
different setting for use when the digital input signal is selected.
When changing the input selection between analog and digital
the required reference level setting is automatically selected. So it
is very easy to optimize levelling and headroom of the model d02
for different applications in analogue or digital mode.

For a digital mastering and transmission the output level should be
the maximum, i.e. the digital reference level should be 0dBFS.

When working with analog inputs it is very important not to overload the A/D
convertor (ADC), in order to ensure that the ADC always provides accurate
linear conversion of the analogue input signal to the digital audio signal which
is used for internal processing.
The analog input gain of the d02 should be set so that the maximum possible
studio output level which will occur in practice must not overload the A/D
converter.
When using the analogue output the analogue output gain following the digital
to analogue converter must also be adjusted so that the internal digital
reference level (maximum digital level which can be output by the digital limiter)
corresponds to the maximum analogue level desired for the recorder or
transmission line. Input a continuous signal such as a tone which is large
enough for the limiter to start to operate and for the maximum output level to be
output. The level on the d02 output level meter should correspond to the
internal digital reference level which was set. Then adjust the analogue output
gain to get the desired maximum analogue output level.

The calibration of the reference level should meet the maximum level of the
transmission line or the transmitter. The internal reference level (limiter
maximum output level) is always the absolute maximum level which the d02 will
output.

12

CONTROL AND DISPLAY
ELEMENTS

All functions of the digital dynamics processor d 02 are
activated by buttons. The front panel shows easily recognizable
function groups.

3

input

By pressing the left button in the input section the required input signal
can be selected. Each time the button is pressed the input selection is
changed and one of the three LED's above the button lights to show
the newly selected input.

When the AES LED is lit the unit processes the AES/EBU format digital
audio signal applied to its AES/EBU input connector.

When ANALOG INT LED is lit the unit processes the analogue input audio
signal aplied to its analogue input connectors, and the sampling frequency
at which the A/D-converter operates is generated internally.
When ANALOG EXT LED is lit the unit uses the same analogue input
audio signal as when ANALOG INT is selected, but now the sampling
frequency at which the A/D converter operates is determined by the
external word clock or AES/EBU input signal which is fed into the unit.

To the right of the input indicator are three LEDs which shows the sample
rate of the selected input. If a given external digital signal (input signal or
wordclock) has the correct sample rate, the device automatically
synchronizes to that frequency and a yellow light appears on the LED. All
LEDs will blink red if the input signal is lacking or the sample rate is outside
the admissible tolerance range.

With internal synchronisation (ANALOG intern) the sample rate display is
green and the frequency can be changed with the button below.

3. CONTROL AND DISPLAY ELEMENTS

Press the PRESET button to select the one of the four operating
programs of the unit which best corresponds to the kind of audio
programme material which is being processed. Each operating program
has optimum values of dynamic control characteristics (such as attack and
release times etc.) for a different type of programme material.

preset

in stereo mode (loop function) in 2-channel mode (loop function)
1 - universal 5 - universal
2 - pop music 6 - pop music
3 - speech 7 - speech
4 - live 8 - live

To change preset group hold down the display button continuously for a
few seconds and the unit will enter the stereo/2-channel setting and the
internal digital reference level setting mode. The PRESET and the GAIN
display flashes and and the GAIN display shows the digital reference level.

The STEREO/2-CHANNEL mode can now be changed pressing the
SELECT button. With every tip the unit toggles between the selected
program in stereo or 2-channel mode. If you leave this setting function you
can select your working program like described above.

gain

The INCrement and DECrement buttons allow a linear amplification of the
digital input signal. The selection of gain levels takes place in steps of 1 dB
and has a range from -6 dB ... +15 dB. Each time the button is pushed
there is a change of 1 dB. Holding down the INC or DEC button continously
leads to a continuous change in gain until the respective end value is
obtained. When the gain level reaches 0 dB there is a short pause to avoid
negative gain (attenuation) being accidentally activated.

13

3. CONTROL AND DISPLAY ELEMENTS

The expander THRESHOLD can be changed upward or downward with two
buttons and is visible on the LEDs above them. Four expander thresholds (-60
dB, -50 dB, -40 dB, -30 dB) can be selected. The threshold level is related to
the choosed digital reference level.

expander

In the OFF position the expander function is switched off.
The activity of the expander is indicated with a red LED in the display
gain reduction.

The compression ratio is adjusted by pressing the RATIO button and the
currently selected ratio is shown by the lighting of the appropriate LED above
the RATIO button.

compressor

One of four different ratios can be selected (1.1 : 1, 1.3: 1, 1.6 : 1, or 2 : 1).
There is also a compressor off position where the compressor function is
turned off. In this case none of the ratio LEDs will be lit.
Compression is partly achieved by increasing the level of low level signals,
(the lowest of which might otherwise be below the noise floor of the FM
transmission system). The lower the input signal level the higher the
additional gain applied to that input signal by the compression processing will
be. The maximum amount of gain applied to a low level signal can be
adjusted independently of the compression ratio. Press both the RATIO
buttons at the same time until normal gain display will be switched off.

maximum compression
gain

A red LED will light in the compressor gain display which indicates the
maximum value. This value can be changed with the keys INC and DEC in
the range of 2 dB ... 15 dB.

14

3. CONTROL AND DISPLAY ELEMENTS

limiter

The limiter limits the maximum output signal level of the d02 precisely to
the set digital reference level. (see also 2.4., and, for details of setting the
digital reference level, see under "display"). The limiter should be always
active to ensure that output level of the d02 never exceeds the preset
digital reference level.
The LED shows a red warning signal when the limiter is turned off.
The limiter works with a look ahead time (signal delay) of approx. 2 ms.This
delay time is present even when the limiter is turned off.
Two different reference levels can be set, one reference level for use when
using a digital input signal, and another for use when using analogue input.

bypass

In the bypass mode (corresponding LED lits red) the digital signal is passing
unprocessed through the DSP to the output. The signal delay time of approx.
2 ms is also effective in bypass mode.
The bypass function is not a relay bypass and is therefore not effective when
the device is turned off from mains power.

15

3. CONTROL AND DISPLAY ELEMENTS

The two channel LED display has three display modes (input level, output
level and gain change). Press the button in the display section to change
the display mode. The selected display mode is indicated by the lighting of
the appropriate LED above the display button and to the left of the display
meters. For better visibility each display mode has its own LED colour and
level meter colour.

display

Green shows the input level and yellow the output level. The scale located

between the two bars indicates the levels. The display which ranges from -50
... 0 dBFS (dB Full Scale) refers to the digital reference level, with a
resolution of 2 dB in the upper section. This does not allow a precise
adjustment, but it does give an indication of the existence and the level of
digital input and output signals.
A peak hold function is available for input and output which makes improved
registration of a momentary peak level possible.
If excessive level at the input occurs when the input level display is selected
(if digital audio samples at the maximum permissable positive or negative
sample value occur at the digital input) then the red clip-LED at the
extreme right-hand end of the level meter lights up and indicates overloads
which are already present in the input signal.
When viewing the OUTPUT level the clipping LED does not light since
the limiter is ON and ensures that the maximum output signal level can not
exceed the preset reference level.
The level meter display is a digital meter without integration time, and
records every successive digital sample value.

The third display mode, gain change, shows the current control levels of
the limiter and compressor in dB.
The compressor works to reduce overall dynamic range by insertion of
additional gain for lower level signals (ie no gain reduction). The scale
above the upper meter bar shows the additonal gain inserted by the
compressor. Lighting of LED's in the meter starts on the left and moves
torwards the right as more additional gain is applied.
The limiter works to reduce the level of high level input signals so that they
do not exceed the preset reference maximum level. The scale below the
lower meter bar shows the level reduction by the limiter. Lighting of LED's
in the meter starts on the right and moves torwards the left as the amount of
level reduction (limiting) required increases.
A red LED is visible in the compressor gain display , which indicates the
maximum permissable value of compressor gain. This value can be
changed in the range +2dB to +15dB (see section on operation of the
compressor on page 10 for details of how to change the maximum
permissable compressor gain).

16

3. CONTROL AND DISPLAY ELEMENTS

17

The DISPLAY button has a second function in addtion to changing the
display mode. It is used for setting the internal digital reference level, which is
the maximum output level which the limiter will allow to be output by the unit.
Hold down the display button continuously for a few seconds and the unit will
enter internal digital reference level setting mode. The GAIN display flashes
and shows the digital reference level.

Setup selections
using display key

The maximum output level permissable for the unit (internal digital reference
level) can now be changed in 1dB steps within the range -15dBFs to 0dBFs
by pressing the INC and DEC buttons. The reference level to be used when
using the analog input and the reference level to be used when using the
digital input can be set independently.

18

FUNCTIONAL DESCRIPTION

After switching the power on, the digital dynamics processor
d02 automatically chooses the settings used before the power

was turned off.
All parameters used, e.g. input, preset, gain, compressor,
expander and display, are stored and re-applied.

The only
exception is the limiter which, as a safety function, is always activated
when the power is switched on.

4

Power-on Setting

The device is capable of processing digital audio signals as well
as analog audio signals. The unit accepts an AES/EBU format
digital audio signal. In the case of a digital audio input signal
being processed the internal sampling frequency of the unit is
automatically synchronised to that of the digital input signal.
The sampling frequency may be any frequency in the range
30KHz to 50KHz. The d02 directly measures the actual
sampling frequency of the AES/EBU input signal with a
frequency counter. It does not rely on the indicated sampling
frequency of the AES/EBU input signal, which is contained in
the signals "channel status" data, being correct.

Digital input signals

If the measured input signal sampling frequency is one of the
standard frequencies (32kHz, 44.1kHz or 48kHz) then a
corrresponding LED will light yellow in the input section on the
units front panel. Continuous lighting yellow of an LED also
indicates that the digital input signal is a valid AES/EBU
digital audio signal which the d02 can synchronise to properly.

Digital input signals

- sample frequency

If AES digital input is selected but the d02 can not
synchronise properly to a supplied AES/EBU input signal (for
example because there is no valid input signal or because the
input signal has a sampling frequency outside the
admissable tolerance range) then all three "sampling
frequency" LED's in the input section of the d02 front panel will
flash red. Digital audio input signals in the standard AES/EBU
format pass from the AES/EBU input connector through a
transformer (as specified by the AES/EBU standard) to the
AES/EBU interface circuitry. The AES/EBU input circuitry
derives the d02's internal sampling frequency from the
AES/EBU input signal and seperates the audio data in the
AES/EBU bit stream from additional control bits, such as
channel status data bits (C-bit) and user bits (U-bit). The audio
sample data is converted from AES/EBU format into the

Digital input signals

- AES/EBU

4. FUNCTIONAL DESCRIPTION

d02's internal digital format for processing. Data in AES/EBU
control bits (C-Bit, U-Bit) will be passed from the AES/EBU
digital input to the AES/EBU digital output unchanged.

The processing of digital audio data in the consumer format
S/PDIF is also possible. If signals are present at both the
AES/EBU and the S/PDIF inputs at the same time, the AES
signal automatically has priority.

Analog audio input signals can be fed into the unit via the analog
input XLR connectors and first pass through an electronically
balanced analog input amplifier, then to an Analog to Digital
converter (ADC). The gain of the electronically balanced
analogue input amplifier can be adjusted, using
potentiometers on the rear panel. The maximum analog
input level , which will correspond to a digital full scale (0dBFs)
digital output signal from the internal ADC, can be set to any
value in the range +12dBu to +22dBu.

The standard factory setting of the unit when supplied is that a
programme level of +6dBu at the analog input corresponds
to -9dBFs (9dB below maximum possible digitally represented
level which can be output by the A/D convertor). Therefore the
maximum permissable analog input level without clipping when
the unit is supplied is +15dBu.

Both analog inputs are converted into digital audio signals, which
can then be processed by the internal digital dynamics
processing. Conversion is done by a high performance 24 bit
oversampling A/D converter which is manufactured by CRYSTAL
Semiconductor. The analogue to digital converter has a
dynamic range of 114dB and is very linear in terms of both
frequency and phase response. Provided that the maximum
permissable analog input level (which will correspond to 0dBFs
(full scale) internal digital input level shown on the units level
meters) is not exceeded the A/D conversion process should
have no signficant influence on the sound quality. The audio
sample data output from the A/D converter is converted into
the d02's internal digital format for processing.

When the input is set to 'ANALOG intern' the sampling
frequency used for the A/D conversion, internal digital
processing and digital output will be generated internally. The
sampling frequency (32KHz, 44.1KHz or 48KHz) can be
selected with the button in the input section and is displayed
by the lighting of a green LED in the d02's front panel sampling
frequency display. For applications where analog input is used,
but where the AES/EBU digital output of the unit must be

Digital input signals

- S/PDIF

A/D-converter

Sample rate

19

4. FUNCTIONAL DESCRIPTION

synchronised with another AES/EBU digital audio signal or with
a Word Clock signal, the input selector must be set to ANALOG
extern.

The AES/EBU signal or Word Clock signal which the unit is
required to synchronise the sampling frequency of its
AES/EBU output with can be applied to the AES/EBU input
connector or to the EXT SYNC word clock input connector
respectively. The sampling frequency of the external
AES/EBU or Word Clock signal (and hence the operating
sampling frequency of the d02) will be indicated by the lighting
of a yellow LED in the d02's front panel sampling frequency
display.

All three LED's in the d02's front panel sampling frequency
display will flash red if the d02 can not synchronise to an
external sampling frequency signal because no signal is
connected or because the connected signal is outside the
admissable working sampling frequency range of the unit
(30KHz to 50KHz).

The digital audio signal (either an AES/EBU digital input signal
or an analog input converted by the A/D converter) is processed
in a Texas Instruments Floating Point Signal processor with a
data width of 32 bits. The use of 32 bit digital audio sample
length in calculation ensures that there is no deterioration in
signal quality, even if AES/EBU digital audio data with the
maximum word length of 24 bits is input into the unit.

The DSP carries out the functions of the dynamics processing,
the linear gain and the emphasis filtering. It measures the input
and output levels and generates data for GAIN CHANGE display.
Reading of the front panel buttons and operation of the front
panel display is performed via a special interface (see also
chapter 3.).

One main task of the digital transmission processor d 02 is the
compression of low signal levels. The compression- RATIO
expresses the effects of a change of the input signal in dB on the
change of the output signal in dB.
E.g. a ratio of 2:1 means that a change in input signal of 20 dB
causes a change in output signal of 10 dB. With the choice of a
compression ratio, the intensity of the compression is determined
and with it also a certain compression characteristic (see also
fig.2 and fig. 3). The RATIO parameter is adjusted on the front
panel in four steps, from 1.1:1 to 2.0:1. The transition to another
characteristic can be carried out during the running programme.
It does not cause any clicking noises.

External
synchronization

Digital signal processor

compression

20

4. FUNCTIONAL DESCRIPTION

The lower the signal level, the higher the gain of the compressor
will be. Independently of compression ratio, the maximum
amount of compression gain can be adjusted so that no
inadmissible increase of background noises (e.g. live atmos, air­conditioning, hum and noise) may occur during signal pauses.
To set the maximum compression gain press both the RATIO
button and the PROGRAM button simultaneously. A red LED
becomes visible In the compressor gain display , which indicates
the maximum value of compression gain. This value can be
changed in 1dB steps over the range +2dB to +15dB pressing
the INC and DEC buttons.

The expander becomes effective when the signal level falls
below an adjustable expander threshold. It is possible to select
four thresholds from -60 dB...-30 dB.
If the level falls below the threshold , the gain is steadily
decreased up to -15 dB. The downward regulation of the
expander is achieved just as quickly as the upward regulation of
the compressor, thereby compensating the resulting increase in
signal noise.

For the dynamics functions, particularly the algorithm of the
limiter, a signal delay of approx. 2 ms is built in. This delay
makes it possible to arrange the algorithm of the limiter in such a
way that the control mechanism is activated before maximum
level is reached. Within the rise time of the signal the peak level
is recognised and the maximum is calculated in such a way that
full scale level is reached precisely without causing clipping.

The processing of digital audio signals in the signal processor
requires a machine-specific format. Special interface circuits are
therefore available to convert to standardised digital interface
formats.

Additionally, an analog output signal is available. A stereo - D/A
converter with a resolution of 20 bits generates an analog signal
with very high audio quality. This signal is fed to balanced output
drivers.The gain of the balanced analog output driver circuit for
each analog output can be adjusted on the rear panel, so that
the maximum possible analog output level can be adjusted to
be any value in the range from +6dBu to +22dBu.

(The maximum possible analog output level here is the analog
output level when the output level meter shows 0dBFs full scale
digital level and the D/A converter is being fed with a digital
signal at 0dBFs - the maximum possible full scale level that can
be represented digitally).

Maximum compression
gain

expander

Look ahead limiter

D/A-converter

21

4. FUNCTIONAL DESCRIPTION

If the internal digital reference level is reduced to below 0dBFs
then the maximum analog output level will be correspondingly
reduced by the action of the limiter.

For example if the internal digital reference level is reduced
to -9dBFs (9dB below the maximum possible level that can be
represented digitally) then the actual maximum digital level
that will be received by the D/A converter will be 9dB below
the maximum possible digital level. In this case the range of
adjustment for maximum analogue output level will be -9dBu to
+13dBu.

The design of the electronically balanced analog output
drivers is such that the output level is maintained even when
driving an unbalanced load.

22

23

APPLICATION NOTES

5

5.1.

Presets

It is possible to choose one of eight different control
characteristics for the dynamics processor. Each of the four
different sets of control characteristics provides ideal dynamics
control for a different type of programme signal as follows:

stereo mode

2-channel-mode
1 - universal 5 - universal
2 - pop music 6 - pop music
3 - speech 7 - speech
4 - live 8 - live

Selecting a particular preset sets up the optimum parameters of
the dynamics processor (attack and release times, threshold
levels and interactions between the multiple signal dependent
control circuits) for a particular kind of programme material.
For example, generally speaking, release times are longest
when using the classical setting and shortest when using the
speech setting. (In order to understand the basic Multi-loop
principle of the Jünger Audio dynamics processors please refer
to chapter 1.2).
Fixed presets containing optimised parameters for different
types of programme signal are used because, with the great
number of parameters used and the interactions of parameters
used in different stages of the multiloop system, changing of
individual parameters by the user could cause problems.

5.2.

Processing signals
containing
emphasis

If the audio signal was recorded with emphasis, the additional
information of the digital input signals contains a definite
emphasis-control-bit in the AES/EBU or SP/DIF format. This is
sometimes the case in older recordings because it slightly
improved the signal-to-noise ratio of currently used analog­digital converters. Similar to noise reduction methods in analog
magnetic tape recording, the higher signal frequencies are
raised prior to recording, and subsequently lowered in playback,
causing a lowering of the higher frequency noise level.

5. APPLICATION NOTES

If such a signal is compressed or limited in a dynamics
processor, problems will occur as the peak levels for high
frequencies do not represent the true values. The dynamics
processor causes a change in peak levels which would, however,
lead to a change in the treble content after passing through the
external deemphasis filter.

Prior to dynamic processing a signal recorded with emphasis
must therefore be linearized, i.e. pass through a digital
deemphasis filter. This filter in the d 02 is automatically switched
on if the corresponding control bit is set in the AES/EBU or
S/PDIF format. If the filter is turned on, the colour of the AES
input LED will change to red.

The static characteristics of the d 02 (see also fig. 2) usually
refers to the digital reference level 0 dBFS (dB Full Scale). This
is useful for most applications of the dynamics processor as the
on-following digital recording system is supposed to be balanced
down to the final bit.
For applications using headroom the d 02 can be adjusted to
another reference level of 0 ... -15 dBFS in steps of 1 dB. The
limiter threshold and therefore the maximum output level are
determined by this digital reference level. This value is then also
the reference for the expander and limiter threshold values. The
static characteristics for a reference level of -9 dBFS are
illustrated in fig. 5.

The adjustment of the device to this reference level is achieved
with pushing DISPLAY and GAIN buttons at the same time (see
also chapter 2.4. and 3.).

5.3.

Working with
headroom

24

5. APPLICATION NOTES

Headroom 9 dB

2.0 : 1

1.6 : 1

1.3 : 1

1.1. : 1

off

static characteristics: compressor

compression gain: max. 10 dB
parameter: ratio
digital reference level: -9 dBFS

-50 -40 -30 -20 -10 0

-50

-40

-30

-20

-10

0

input level (dBFS

)

output level d02 (dBFS

)

The audio signal delay through the dynamics processor is
approx. 2ms due to delaying of the audio signal using internal
memory. A small delay is deliberately introduced to the audio
signal in order to allow limiter and compressor algorithms which
can 'preview' the audio signal before changing it. That is the
signal curve can be changed before maximum level is reached.
(For further details see chapter 1).
This delay must be considered before attempting to mix
signals processed by the dynamics processor with other
undelayed signals.

When mixing together a delayed signal and a direct signal
there may be cancellation of the signal waveform at some
frequencies and re-inforcement of the waveform at other
frequencies (comb filter effect). Corresponding 2ms delay of
direct signals should therefore be carried out before mixing
them with delayed processed signals.

fig. 5:
Static
characteristics:
Compressor/
Limiter with -9dBFS
Digital Reference
Level

5.4.

Influence of signal
delay time

25

5. APPLICATION NOTES

5.5.

Selection of
parameters to
increase loudness

Signal compression and the loudness enhancement of the digital
audio signal resulting from it can be achieved by combining two
dynamic range control processes: firstly, the compression
achieved by increasing small and medium signal levels and
secondly, linear amplification combined with the inaudible
limitation of individual, remaining peak levels with the limiter.

In the gain change mode the operation of compressor and limiter
can be observed on the display. For smaller signal levels the
compressor causes additional amplification which however
decreases the higher the signal level is . With full scale levels the
compressor is practically ineffective so that even an increase of
the RATIO will have no effect.
If you now increase the linear amplification GAIN, individual peak
levels are raised above the limiter threshold and limited inaudibly.
All other signal components can however be increased. If the gain
is too large also medium levels are treated by the limiter, which
means that the limiter then reduces the signals continually and
again reduces the additionally applied amplification.
The display for Limiter-Gain-Reduction should be in the region of

0...-6...-8 dB and should not light up red continuously, so that a
dynamic limitation only applies to signal peaks. Then the signal
compression and therefore also the increase of loudness is at its
most effective.

5.6.

Redithering ­Reduction of word
length of digital
output signal

At the end of postproduction the material must be prepared for
copy on COMPACT DISC. The information of a 24 bits signal is
not more storable linear. One must shorten 24 bits data word to 16
bit word length. The practice offers several procedures for it.

In the simplest case, the last bits are cut off - truncation. One
requires no further processing to this, it is enough to record a 20or
24 bits signal direct on a 16 bits storage medium. In this case, a
not unimportant quantization mistake however results, the part of
the harmonious distortions increases. Single numeric roundoff of
the signal to 16 bits reduces this mistake. Nevertheless, the result
will normally be worse than the data by the same original analog
signal converted with a good 16 bit ADC.

truncation

26

5. APPLICATION NOTES

27

In order to receive a better quality during cut down the data to 16
bit one must redithering the material with corresponding devices.
Here the device is calculating random numbers (dither signal) and
add a different random number to every sample. Then it will be cut
off to 16 bit. As a result, the bit with least weight (LSB) is put in
such a way that it corresponds best to the information of the last
bits following available ones no more and makes less distortions
as hissing in the signal.

redithering

A specific redithering is noise shaping, with that the noise
modulation of the LSB is considering the psycho-acoustic
sensitiveness of the human ear. With it can be enlarged the
audible dynamic range of a reformatted 16 bit recording ideally by
about 3 bits i.e. to more than 110 dB. To make this quality audible,
the CD must of course be played back with appropriate
monitoring equipment (D/A converter, amplifier) for 20 bit quality.

noise shaping

The disadvantage of redithering - noise shaping consists in the
restrictions during postproduction. In order to receive the achieved
effect every processing must occur with coefficients which
correspond to the word length of the initial data. At signals
processed with noise shaping, these coefficients would have to be
besides filtered in the same manner like the dither signal. If one
can not adhere to these conditions, one must live with the loss of
the effect, in some cases it occurs data losses. Multiple
application of these procedures can even make drop-in and
noises, such as twittering or clicks. Therefore, noise shaping or
redithering should be used only at the end of the process chain,
i.e. during the preparation of the copy master for reproduction.

Disadvantages of
noise shaping/
redithering

7

APPLICATIONS

6

- mastering of CD, DCC, MD

maximum recording level without clipping

increased programme density and loudness

- digital recording and mixing

increased loudness level (compressor, limiter)

eliminating noise signals (expander)

- FM-Broadcast, TV-Sound

signal conditioning
matching dynamic range of different programme signals
increasing signal loudness level

- limiter for digital or analog transmission links

always digital full scale signal, without clipping

- post production and ADR studios

adjusting dynamic range and loudness level of individual takes,
maximum recording level without clipping

- A/D converter free of overload for general use

high performance 24 Bit ADC in combination with digital limiter

digital output signal without clipping

further applications without the dynamic functions

- digital audio format conversion

all digital outputs are available in parallel
irrespective of the input format
AES/EBU + S/PDIF + OPTICAL

- digital deemphasis filter

removing emphasis automatically
emphasis bit in AES/EBU is also removed

- digital-analog converter

high quality 24-bit stereo output signal
balanced line outputs with adjustable output level

TECHNICAL
SPECIFICATION

29

sample rate : 30 kHz ... 50 kHz
audio data format : 24-bit (AES/EBU)
24-bit (A/D-,D/A-converter)

7

AES/EBU
level : 5 Vpp / 110 Ohm, balanced
connector : XLR
input format : AES professional, AES consumer
output format : same as input

S/PDIF
level : 0.5 Vpp / 75 Ohm, unbalanced
connector : RCA
input format : AES professional, AES consumer
output format : same as input

OPTICAL
connector : TOSLINK

A/D-converter : stereo, 24 Bit, oversampling
dynamic range : 112 dB (RMS)
114 dB (A-weighted)
input level : +12...+22 dBu for 0 dBFS, adjustable
input : XLR, floating balanced, 10 kOhm
(optional: transformer balanced)

D/A converter : stereo, 24-bit, oversampling
dynamic range : 108 dB (RMS)
110 dB (A-weighted)
output level : +12...+22 dBu for 0 dBFS, adjustable
output : XLR, floating balanced, 50 Ohm
(optional: transformer balanced)

digital
input / output

analogue
input / output

general

remote : for connection with d - remote drc01
(optional)
power consumption : approx. 20 W
dimensions : 19 inch, 1 RU, 250 mm depth
weight : appr. 4.5 kg

30

WARRANTY AND SERVICE
INFORMATION

8

JÜNGER AUDIO grants a two-year warranty on the

digital dynamics processor MODEL d02

If the unit has to be serviced, please send it, ideally in the
original box, to:

JÜNGER AUDIO - Studiotechnik GmbH

Justus-von-Liebig-Str. 7

D - 12489 Berlin

GERMANY

Tel.: (*49) -30-677721-0

Fax.: (*49) -30-677721-46

Geräteart: Digital Filter Processor

KONFORMITÄTSERKLÄRUNG

DECLARATION OF CONFORMITY

Type of equipment: digital filter processor

Produkt

/ Product: model d02

Das bezeichnete Produkt stimmt mit den Vorschriften folgender
EU-Richtlinie(n) überein:

The aforementioned product complies with the following Europaen Council Directive(s):

89/336/EWG (geändert durch 91/263/EWG und 92/31/EWG)

(changed by 91/263/EWG and 92/31/EWG)

Richtlinie der Rates zur Angleichung der Rechtsvorschriften der

Mitgliedsstaaten über die elektromagnetische Verträglichkeit
Council Directive 89/336/EC on the approximation of the laws of the
Member States relating to electromagnetic compatibility

Zur vollständigen Einhaltung dieser Richtlinie(n) wurden folgende Normen
herangezogen:

To fully comply with this(these) Directive(s), the following standards have been used:

EN 55022 :1987
EN 50082-1 :1993

Dieser Erklärung liegt zugrunde: Prüfbericht(e) des EMV-Prüflabors

This certification is based on: Test report(s) generated by EMC-test laboratory

MEB Messelektronik Berlin Kalibrier- und Prüflabor

accredited EMC laboratory

Aussteller

/ Holder of certificate: Jünger Audio Studiotechnik GmbH

Justus-von-Liebig-Str. 7
D - 12489 Berlin

Berlin, 02.11.1995 Herbert Jünger

(Ort/Place) (Datum/Date) (Rechtsgültige Unterschrift/Legally Binding)

CE Konformitätserklärung Jünger Audio Studiotechnik GmbH, Berlin Nobember 1995