Junger Audio b46 User Manual
4ch digital audio leveller
b46
™
LEVEL MAGIC release 3.0
LM2
FOREWORD
Thank you for buying and for using the 4-channel Digital Audio
Level Processor b46.
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 Audio Level
Processor b46.
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 b46, 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 Audio Leveller Level Magic ™ ………….
2.3.3 Limiter ……………...................................
2.3.4 Transparent mode ................................
2.4 The Jünger Audio Dynamics Principle ..........
2.4.1 Program …………………………...........
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 audio group ........................
2-1
2-1
2-2
2-3
2-3
2-3
2-5
2-5
2-5
2-6
2-7
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
5. Operation ...............................................................................
5.1 adjustment of parameters …………………………………..
5.2 gain display ……..……………………………………………
5.3 mode menu …………………………………………………
5.4 input menu ……………………………….…………………
5.5 leveller menu …………………..…………………………….
5.6 limiter menu …………………………………………………
5.7 utility menu …………………………………………………
5.8 recall and storage of presets ……………………………..
5.9 editing of presets …………………………………………
5.10 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
5-3
5-3
5-4
5-6
5-7
5-8
5-8
5-10
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 b46 is a professional studio
device that is performing automated levelling of digital audio
signals.
The dynamic range processor principles developed by Jünger
Audio enable level managing devices like compressors, AGC
and limiters 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 in combination with adaptive
controlled processing algorithms 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 audio levelling 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 audio levelling processing
• 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
AGC, Transient Processor, Limiter
2
2.1
BASIC
DESCRIPTION
operation manual b46, chapter 2 -Function description- page 2-1
2. FUNCTION DESCRIPTION
2.2
BLOCK DIAGRAM
page 2-2 operation manual b46, 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 signals. The input gain can
be changed in steps of 0.1 dB , within a range from -15...+15 dB.
Adjustment of GAIN is channel independent.
Level Magic ™ is a unique algorithm to make automated audio levelling
possible.
Input level change
Pic. 2 is showing a
theoretical level change
of +5dB and –5dB
around program level.
Working with AGC
In pic.3 a conventional
AGC is used to adjust
the level. As we can see
the AGC needs a certain
time to react, that is
necessary for mostly
inaudible gain correction.
But that’s too long to get
a proper correction of the
input level change.
Level Magic ™
Level Magic ™ is a
unique combination of a
transient processor and
an adaptive AGC
process. The transient
processor can fill the
lack of level control
against the slow acting
AGC. The total gain of
Level Magic ™ is the
addition of the gain by
the transient processor
and the gain of the AGC.
Transient
Processor
AGC
adaptive
AGC
2.3
AUDIO SIGNAL
PROCESSING
2.3.1
GAIN
2.3.2
AUDIO LEVELLER
LEVEL MAGIC ™
operation manual b46, chapter 2 -Function description- page 2-3
2. FUNCTION DESCRIPTION
Adjustment
procedure
Process
description
Parameter
description
The Level Magic ™ process needs to be setup in three steps
- select one of the default presets for your apllication
(see preset description in chapter 5)
- adjust the operation level and peak level referring to
standards that are using for your application
- if the default preset is not giving satisfying results change
the parameters indivdually
Level Magic ™ is using a unique combination of QP and RMS level
detectors to analyze the incoming audio signal. In comparing QP and
RMS measurement results we can find out how much transients are
coming in. Dependent on that the necessary resulting gain is controlled
in relation between transient processor and AGC.
Limiter
Transient
Processor
Level
Detection
AGC
Transient processor is doing fast gain change and the AGC is doing
slow gain change (depending on settings). The way how Level Magic is
acting on the audio is mostly determined by balancing between slow
and fast gain changing process. The AGC should be set in a way that
the gain change is mostly inaudible (1dB per 5 seconds or slower). The
Transient Processor should be set that incoming level jumps are
reduced but originally dynamic range is not changed too much. As
more possible gain by the Transient processor as more reduction of
the dynamic range is coming with.
SOFT level control: AGC range …15dB, time >=2min
Transient range …4dB, soft process
MID level control: AGC range …12dB, time >=1min
Transient range …6-8dB, mid process
HARD level control: AGC range …10dB, time >=40sec
Transient range …10dB, hard process
Parameter description:
AGC
OP-level operation level, target level for the AGC and for
the Transient Processor
Range max. gain by the AGC
Time time to reach the max. gain change
Gate threshold level that the AGC stops dynamic gain
change and is moving gain slowly to the long
term average gain change value
Transient Processor
Process a combination of level ratio and release
characteristic for the fast gain change (soft, mid,
hard)
Range max. gain by the Transient Processor
page 2-4 operation manual b46, chapter 2 -Function description-
2. FUNCTION DESCRIPTION
The static characteristics of the b46 limiter 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) at a limiter threshold of 12dBFS are illustrated in fig. 6
.
limiter
threshold
0...-20dBFS
input
-60
max. output
level
[dBFS]-10-20-30-40-50
-10
-20
2.3.3
LIMITER
fig. 6:
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.
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
2.3.4
TRANSPARENT
MODE
2.4
THE JÜNGER AUDIO
DYNAMICS
PROCESSOR
PRINCIPLE
operation manual b46, chapter 2 -Function description- page 2-5
2. FUNCTION DESCRIPTION
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.
A A
delay
1
1
2
n
Multi - Band
f
2
m
Multi - Loop
f
2.4.1
PROGRAM
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.
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).
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
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
page 2-6 operation manual b46, chapter 2 -Function description-
2. FUNCTION DESCRIPTION
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
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.
2.4.2
INFLUENCE OF
SIGNAL DELAY
TIME
operation manual b46, chapter 2 -Function description- page 2-7
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