Junger Audio b42 User Manual
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jungeraudio
b42
FOREWORD
Thank you for buying and for using the 4-channel Digital
Dynamics Processor b42.
Not only you have aquired the latest generation of digital
dynamic range processing, but also a piece of equipment which
is unique in its design and specification.
Please read this manual carefully to ensure you have all the
information you need to use the 4-channel Digital Dynamics
Processor b42.
The unit was manufactured to the highest industrial standards
and went through extensive quality control checks before it was
supplied.
If you have any comments or questions about installing, settingup or using the b42, please do not hesitate to contact us.
0
The information contained in this manual is subject to change without notice.
This manual is the copyright of Jünger Audio. All reproduction and copying, other than for legal owner’s
personal use, or disclosure of part or whole to a third party, is not allowed without prior written
authorization by Jünger Audio. © Jünger Audio, Berlin 1998-2004
1
CONTENTS
2. Function description .......................................….
2.1 Basic description ...........................................
2.2 Block diagram .............................................
2.3 Audio signal processing ................................
2.3.1 Gain ......................................................
2.3.2 Expander …………………………………
2.3.3 Deesser ………………………………….
2.3.4 Compressor ……………………………...
2.3.5 Limiter ……………...................................
2.3.6 Program …….. .......................................
2.3.7 Transparent mode ................................
2.4 The Jünger Audio Dynamics Principle ..........
2.4.1 Selection of parameters to increase
loudness ….................................................
2.4.2 Influence of signal delay time …............
3. Installation .........................................................
3.1 Unpack the unit ...........................................
3.2 Power supply ................................................
3.3 Connections ...........................…...................
3.4 Rack mounting ..........................…................
3.5 Operation safety ..........................……..........
3.6 Synchronization of digital output ..….............
3.7 Remote Control ..................……...................
3.7.1. GPI Remote Control ......…….............
3.7.2. Tally Out .............………....................
3.7.3. Serial Remote Control .......…….........
4. Location of parts and controls ............………......
4.1 Front panel ......................……......................
4.2 Rear panel ...........................………..............
4.3 Switches and jumpers for configuration ........
4.4 Selection of SDI Split Mode ......…….............
4.5 Selection of SDI audio group ........................
2-1
2-1
2-2
2-3
2-3
2-3
2-3
2-4
2-5
2-5
2-6
2-6
2-7
2-8
3-1
3-1
3-1
3-1
3-1
3-1
3-2
3-3
3-3
3-4
3-5
4-1
4-1
4-3
4-4
4-5
4-5
5. Operation ...............................................................................
5.1 adjustment of parameters …………………………………..
5.2 gain menu ……………………………………………………
5.3 mode menu …………………………………………………
5.4 expander menu ………………………………………………
5.5 de-esser menu ………………………………………………
5.6 compressor menu …………………………………………..
5.7 limiter menu …………………………………………………
5.8 utility menu …………………………………………………
5.9 recall and storage of presets ……………………………..
5.10 editing of presets …………………………………………
5.11 list of factory presets ……………………………………..
6. Boot display and trouble shooting ............................................
6.1 Boot display .......................................................................
6.2 Error messages and trouble shooting ................................
6.3 Initialization the unit ............................................................
7. Application notes ......................................................................
7.1 B40 series with SDI interface .............................................
7.2 Basic working modes with SDI ...........................................
8. Technical specifications ...........................................................
9. Warranty and service information ...........................................
5-1
5-2
5-2
5-2
5-4
5-4
5-4
5-5
5-6
5-6
5-7
5-8
6-1
6-1
6-1
6-2
7-1
7-1
7-1
8-1
9-1
2. FUNCTION DESCRIPTION
FUNCTION DESCRIPTION
The digital dynamics processor b42 is a professional studio
device that processes the dynamic range of digital audio
signals.
The dynamic range processor principles developed by Jünger
Audio enable compressors, limiters and expanders to be
produced with exceptionally high audio quality, without
coloration, pumping, breathing, distortion or modulation effects
sometimes associated with this type of processor.
In short, almost inaudible processing - with ease of use. The
outstanding quality of the processing is based on the Multi-Loop
dynamic range control principle developed by Jünger Audio.
The unit is easy to operate and requires only a limited number of
settings to be made by the user to achieve optimum results. All
other parameters necessary for inaudible processing are
continuously automatically controlled in response to changes in
the programme signal.
features
• 4-channel digital dynamics processor
• various link modes: 4-ch, stereo 1/2 or 3/4, ch1...4
independent
• adjustable input gain (channel independent) -15...+15 dB
• adaptive controlled dynamic range processing
expander on/off, THRS -50...-20 dBFS, REL ...
de-esser on/off, TYPE male/female, RNG -20...0 dB
compressor on/off, RATIO 1,0:1...4,0:1, RNG 0...15 dB
limiter on/off, THRS 0...-20 dBFS, PROgram 1..4
• user friendly preset and recall function (10 presets)
• pairwise bit transparent mode input to output
• extern sync mode, AES/EBU or VIDEO (or SDI if
optional SDI-interface is present)
• RS-422 interface for serial remote
• GPI interface for parallel remote control, tally output
2
2.1
BASIC
DESCRIPTION
operation manual b40, chapter 2 -Function description- page 2-1
2. FUNCTION DESCRIPTION
2.2
BLOCK DIAGRAM
page 2-2 operation manual b40, chapter 2 -Function description-
2. FUNCTION DESCRIPTION
All signal processing is done in the digital domain by Texas
Instruments floating point signal processors. The use of 32 bit word
length for calculation ensures that there is no deterioration in signal
quality, even if an audio signal with a maximum word length of 24 bit is
input into the processing of the unit.
GAIN means linear amplification of input or output signals. The input or
output gain can be changed in steps of 0.1 dB , within a range from -
15...+15 dB.
Adjustment of GAIN is channel independent.
Below an adjustable threshold level an expander can be activated
which can lower the amount of noise signals.
input
threshold
level
-20...-50, off
OFF
[dBFS]-10-20-30-40-50-60
-10
-20
2.3
AUDIO SIGNAL
PROCESSING
2.3.1
GAIN
2.3.2
EXPANDER
fig. 2:
static
characteristics:
expander
ratio
1:4.0
range
20 dB
-30
-40
-50
output
The de-esser is a special processing function to reduce Sfrequencies of speakers. This can be done either by using a
compressor with frequency selective side chain, or by dynamic filtering
of voice signals.
The de-esser of the b42 uses a sophisticated dynamic filtering
algorithm for the reduction of S-frequencies. The dynamic filter makes
it possible to reduce these frequencies without influencing other
spectral parts, and works independent of the signal level.
The critical S-frequencies are different for female and male voices.
Only two basic adjustments are necessary for the de-esser - filter
frequency and the amount of s-reduction (range).
All other parameters which are necessary for effective de-esser
function are controlled by the audio signal itself.
The threshold of the de-esser is automatically set and follows the
signal power level. The reduction of S-frequencies can be controlled
by setting the range parameter from 0...-20dB.
2.3.3
DE-ESSER
operation manual b40, chapter 2 -Function description- page 2-3
2. FUNCTION DESCRIPTION
fig.3:
basic function:
de-esser
-10
-20
10k1k500100
20k
frequency
auto
threshold
function
2.3.4
COMPRESSOR
fig.4:
static
characteristics:
compressor
S-reduction
range 0...-20
Input
level
[dBFS]
-30
-40
-50
filter
frequency
1..14kHz incl.
male/female
The compression of the programme signal takes place evenly over
the entire input level range and not only at the upper end above a
certain threshold level. Dynamic structures of the input signal (e.g.
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.
input
ratio
2.0:1
[dBFS]-10-20-30-40-50-60
-10
1.6:1
1.3:1
OFF
range
max. 15dB
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.
page 2-4 operation manual b40, chapter 2 -Function description-
-20
-30
-40
-50
output
2. FUNCTION DESCRIPTION
Independent of the compression ratio , the gain of the compressor
(range) can be limited (maximum 1 dB to 15dB), so that there should
be no inadmissible increase of background noises during signal
pauses (e.g. live atmos, air-conditioning, hum and noise).
The static characteristics of the b42 usually refers to a digital output
level of 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 output level of can be adjusted
within 0 ... -20 dBFS in steps of 0.1 dB. The limiter threshold
determines the maximum output level.
The static characteristics fo limiter (solid) and compressor (dotted) at a
limiter threshold of -12 dBFS are illustrated in fig. 5
.
input
limiter
threshold
0...-20dBFS
audio processing of
compressor is related
to limiter threshold level
max. output
level
[dBFS]-10-20-30-40-50-60
-10
-20
2.3.5
LIMITER
fig. 5:
basic function:
limiter
-30
-40
-50
output
For the dynamics functions 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 (look ahead limiter). 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.
For some of the control parameter it is possible to define a limited
range of time constant values which is allowed for the adaptive
dynamic range algorithms. Inside this range the time constants can be
varied by the adaptive processing. Setting the range of time constant
values may be sometimes useful, to get the best signal processing
performance regarding specific programme material.
Parameter related to the transient response of the control circuit are
important for distortionfree processing. These time constants are
allways adaptive controlled without remarkable limitation of parameter
range. This is caused by the presence of transient pulses in allmost
2.3.6
PROGRAM
operation manual b40, chapter 2 -Function description- page 2-5
2. FUNCTION DESCRIPTION
2.3.7
TRANSPARENT
MODE
2.4
THE JÜNGER AUDIO
DYNAMICS
PROCESSOR
PRINCIPLE
each kind of programme material. The algorithm has to guarantee best
reaction for fast increasing level of transient signals anytime even if
classical music with slow dying out characteristic is processed. In all
cases the attack time of the limiter for very short transients is zero.
Especially the release time of the control circuit has more influence to
the increase of loudness as any other parameter. The ranging of time
constants in processing time groups reflects this fact. The range for
processing time shows influence on release time parameter mostly.
The selection of the parameter PROGRAM changes the range of time
constant values as follows:
PRO processing time corresponds to
preset
--------------------------------------------------------------------------------------------- 0 2 ms to 0.2 sec
1 5 ms to 0.5 sec LIVE
2 10 ms to 0.8 sec
3 15 ms to 1.2 sec SPEECH
4 30 ms to 2.5 sec POP
5 50 ms to 3.5 sec
6 70 ms to 5.0 sec UNIVERSAL
7 100 ms to 6.0 sec
8 150 ms to 8.0 sec CLASSIC
9 250 ms to 10.0 sec
In case that the input signal (audio pair 1/2 or/and 3/4) is not audio (but
AC-3, Dolby E, MPEG..) the input can be feeded directly to the related
output bit transparent (no bit changes). The unit is switching to
transparent automatically if “non audio” flag in the Channel Status Bit
of the AES signal is set. Otherwise transparent mode can be set
manually by the user.
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.
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. This is quite different to the popular
multiband structure which changes the sound.
The resulting attack and release times of the Multi-loop-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.
page 2-6 operation manual b40, chapter 2 -Function description-
2. FUNCTION DESCRIPTION
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).
A A
delay
1
1
2
n
Multi - Band
f
2
m
Multi - Loop
f
Signal compression and the loudness enhancement of the digital
audio signal 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
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.
2.4.1
SELECTION OF
PARAMETERS TO
INCREASE
LOUDNESS
operation manual b40, chapter 2 -Function description- page 2-7
2. FUNCTION DESCRIPTION
2.4.2
INFLUENCE OF
SIGNAL DELAY
TIME
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. 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 reinforcement 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.
page 2-8 operation manual b40, chapter 2 -Function description-
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