junger d02 User manual

digital dynamics processor
d02
release 3.0 rev.12

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, 24 bit oversampling DAC
adjustable level, balanced
- redithering for 16 or 20 Bit digital output format

CONTENTS
0
1. The design of the device ..................................................
1-1
1-1
1-1
1-5
1.1. Basic functions ...........................................................
1.2. The Jünger Audio dynamics processor principle .......
1.3. A/D-conversion with digital full scale level ..................
2. Installation .........................................................................
2-1
2-1
2-1
2-2
2-3
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 ............................................
3-1
4. Functional description .......................................................
4-1
5. Application notes .............................................................. 5-1
5-1
5-1
5-2
5-2
5-3
5-3
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 .......................................................................
6-1
7. Technical specification ...................................................... 7-1
8. Warranty and service information ..................................... 8-1

1. THE DESIGN OF THE DEVICE
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. In addition, an analog
stereo signal output is available which operates with 24-bit D/A
converters and enables a rapid acoustic monitoring.
1.1.
Basic
Functions
1
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).
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.
1.2.
The
Jünger Audio
Dynamics
Processor
Principle
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.
1-1

1. THE DESIGN OF THE DEVICE
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.
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.
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 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.
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).
traditional compressor
and limiter designs
multi-band structure
multi-loop principle
delay time
1-2

1. THE DESIGN OF THE DEVICE
A A
f
delay
1
1
2
2
n m
f
Multi - Band Multi - Loo
p
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 s
fig. 1:
basic principles of
dynamic range
processors
compressor
ound 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
1-3

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: com
p
ressor
compression gain: max. 10 dB
parameter: ratio
-50 -40 -30 -20 -10 0
-50
-40
-30
-20
-10
0
in
p
ut level
(
dBFS
)
out
p
ut level d02
(
dBFS
)
off
5 dB
10 dB
15 dB
static characteristics: com
p
ressor
compression
g
ain: max. 15 dB
parameter: compression
g
ain
ratio: 1.6 : 1
-50 -40 -30 -20 -10 0
-50
-40
-30
-20
-10
0
in
p
ut level
(
dBFS
)
out
p
ut level d02
(
dBFS
)
fig. 3:
static
characteristics:
compressor
1-4

1. THE DESIGN OF THE DEVICE
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 analog-digital converter caused
by the necessary headroom of 6 ... 10 dB to prevent over-level of the
signal wich could otherwise occur.
1.3.
A/D-
Conversion 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.
1-5

2. INSTALLATION
INSTALLATION
2.1.
Power Supply
2.2.
Connections
2
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
ANALOG OUTPUTANALOG INPUT
LEFT RIGHTLEVEL
OUT
AES/EBU S/PDIF
OPTICAL
OUT
IN
IN
model d02
230 V
50 Hz
200 mA
Nr
REMOTE
EXT SYN C
16
20
BIT
DIG OUT
24
CON
STATUS
PRO
POWER INPUT
IEC mains input connector 100-240V, 50/60 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)
2-1

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 24bit 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)
2-2

2. INSTALLATION
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.
2-3

3. CONTROL AND DISPLAY ELEMENTS
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.
3-1

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.
in stereo mode (loop function) in 2-channel mode (loop function)
1 - universal 5 - universal
2 - popl 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.
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.
preset
gain
3-2

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.
3-3

3. CONTROL AND DISPLAY ELEMENTS
3-4
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.
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.
limiter
bypass