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.
9. Warranty and service information ...........................................
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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
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.
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.
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
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.
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.
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.
The digital dynamics processor b42 was carefully packed in the
factory and the packaging was designed to protect the
equipment from rough handling. Please examine carefully the
packaging and its contents for any signs of physical damage,
which may have occured in transit.
The digital dynamics processor b42 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.
The dynamics processor b42 is equipped with standard
connectors (see also chapter 3).
Before connecting the digital dynamics processor b42 switch the
power off at all connected units.
The digital dynamics processor b42 is made as standard 19“ unit
(EIA format). It occupies 1 RU (44 mm height) space in a rack.
Please allow at least additional 3“ depth for the connectors on
the rear panel.
When installing the unit in a 19“ rack the rear side of the unit
needs some support, especially for mounting in flight cases.
The digital dynamics processor b42 should not be installed near
units which produce strong magnetic fields or extreme heat. Do
not install the filter processor directly above or below power
amplifiers.
If, during operation, the sound is interrupted or displays no longer
illuminate, or if abnormal odor or smoke is detected immediately
disconnect the power cord plug and contact your dealer or
Jünger Audio.
The digital dynamics processor b42 has a digital signal output
only. To the problem-free combination of following digital devices,
the digital signal processing can be locked to an external clock
reference. The selection of the corresponding input is made in
the SYNC MODE menu. If the chosen sync input is connected
with the sync signal, this signal is used for synchronization
automatically. The digital output signal can be clocked with the
following clock frequencies:
CH 1/2locks with the clock frequency of the input signal at digital input CH 1/2 (AES/EBU, 48 kHz)
EXT SYNC locks with the clock frequency at the
external sync input (AES/EBU, 48 kHz)
VIDEO locks with the clock at the Video sync input
(internal 48 kHz)
SDI VIDEO locks with the clock at the SDI input
(internal 48 kHz)
Both digital outputs CH 1/2 and CH 3/4 are locked with same
clock frequency.
Note: SDI sync is available only if SDI interface is installed!
The digital dynamics processor b42 can be remote-controlled by
means of parallel GPI contacts.
use : remote-controlled changeover of presets
connector: D-SUB 15pin, female
Pin assignments
Pin Signal name Logic I/O Functions
1 PRESET1 H I recall preset1
2 PRESET2 H I recall preset2
3 PRESET3 H I recall preset3
4 PRESET4 H I recall preset4
5 PRESET5 H I recall preset5
6 PRESET6 H I recall preset6
7 PRESET7 H I recall preset7
8 PRESET8 H I recall preset8
9 MUTE H I Muting outputs
10 BYPASS H I bypass on
11 not used
12 not used
13 not used
14 Common pin External ground
15 +5V O Test power source
GPI input potential free by opto-coupler, low activeOFF: +3.5…+30V between GPI input
and pin14
ON: less then 1.5V
Min 50ms
Note: If using an external voltage feed it has to be connected to pin 14!
External Ground is switching the GPI on any of the inputs.
An internal voltage feed is available on pin 15. Ground is available from the
shield of the connector only! By using the internal voltage feed there is no
electrical isolation given anymore.
Pin14
Pin1...13
Ext. voltage
feed+3
Contact to
ext. Ground
5...30V
min. 50 ms
+3,5...30V
+1,5V
3. INSTALLATION
3.7.2
The digital dynamics processor b42 can transmit specific device
statuses via parallel Tally lines.
use: Control of the remote-controlled changeover of
Pin Signal name I/O Functions
1 T1 open contact O preset1 recalled
2 T2 open contact O preset2 recalled
3 T3 open contact O preset3 recalled
4 T4 open contact O preset4 recalled
5 T5 open contact O preset4 recalled
6 T6 open contact O preset4 recalled
7 T7 open contact O preset4 recalled
8 T8 open contact O preset4 recalled
9 T9 open contact O mute
10 T10 open contact O bypass
11 T11 open contact O Not used
12 T12 open contact O Not used
13 T13 open contact O Not used
14 T14 open contact O Not used
15 root Common root contact
Electrical specification:
Tally output type: normally open relais contacts
Contact rating: 1A 24 VDC, 0,5 A 125 VAC
max. 30 W 62,5 VA
The cable is wired 1:1 completely, the shield of the cable must be connected
on both ends!
REMOTE IN
Pin Signal name Functions
1 DSR + out Data set ready
2 DSR - out
3 SENSE in Interrogation Remote
4 RXD + out Receive data
5 RXD - out
6 DTR + in Data terminal ready
7 DTR - in
8 TXD + in Transmit data
9 TXD - in
REMOTE OUT
Pin Signal name Functions
1 DSR + in Data set ready
2 DSR - in
3 GND GND
4 RXD + in Receive data
5 RXD - in
6 DTR + out Data terminal ready
7 DTR - out
8 TXD + out Transmit data
9 TXD - out
input for ext. sync signal (AES 3 format, 75 Ohm, unbal)
input for video sync signal (blackburst, 75 Ohm, unbal)
SDI IN / OUT (only if installed!)
Input/output for serial digital video (ITU-R BT.601, SMPTE 272M-A)
with embedded audio Format: 270 Mb/s, 525/625 line rate, 75 Ohm,
connector: BNC socket
DIGITAL IN
input for AES/EBU standard format
connector: XLR female panel jack
1- ground, 2-3 signal, balanced
connector: BNC socket 75 Ohm, unbalanced
Operation manual b42, chapter 4 -location of parts and controls - page 4-3
4. LOCATION OF PARTS AND CONTROLS
4.3
SWITCHES AND
JUMPERS FOR
CONFIGURATION
Some basic settings are to select by switches on the rear panel
or by switches and jumpers at the internal circuit boards of the
unit. These settings can occur general changes for operation
and should made by qualified engineering staff only.
Rear panel
Selection of the device address for serial
remote, 16 device addresses selectable
device needs a different address! The selected
address is valid after next power-on reset of the unit.
Internal
To set any internal jumper or switches it is necessary to open
the unit.
PLEASE DO NOT MAKE ANY ALTERATIONS WITH THE
MAINS STILL CONNECTED TO THE UNIT!
Loosen the screws on the top cover and remove. Then you can
see all jumper and switches as shown in the drawing below.
After setting of jumper or switches reassemble the unit in
opposite order.
Note: Within a line of remote controlled units every
SDI
Interface
B4x
DSP card
page 4-4 Operation manual b42, chapter 4 -location of parts and controls -
J1
SDI Split
J2
Download
Main board
4. LOCATION OF PARTS AND CONTROLS
Units with SDI interface can be used in SDI split mode:
Audio in path SDI input > AES output
Audio out path SDI output > dsp processing > AES output
(see also 2.5)
External device
SDI
in
AES
in
AES
out
AES
out
AES
in
SDI
out
dsp
B4x
processing
Split Mode
The selection of split mode (SDI DIRECT) is made by switch
SPLIT on in the input (MODE) edit menu.
The 4-channel processors of b40 series fitted with SDI-interface
are compatibel with the standard SMPTE 272M-AB. They
support 48 kHz synchronous audio sampling with 20 bit word
length.
The standard allows up to four groups each of four mono audio channels.
(Usually used by most of D-VTR's and other equipment is Group 1 with 48
kHz synchronous sampling.)
Group selection and other settings are to configure with settings
by front panel operation (mode section).
4.4
SELECTION OF
SDI SPLIT MODE
4.5
CONFIGURATION
OF SDI INTERFACE
Operation manual b42, chapter 4 -location of parts and controls - page 4-5
5. OPERATION
p
p
OPERATION
The use of the digital dynamics processor b42 is very easy.
The setup or the programming of the digital dynamics
processor b42 is made by adjustment of various parameters
and settings.
The description is made related to the functions in the menus.
5.1 adjustment of parameters
5.2 level display
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
Following syntax is used:
SYMBOL ACTIVITY
describes
how to use
button or
rotary knob
ush
turn
ush + turn
describes
action or function of
button or
rotary knob
After selection of one of the utility or function menus by pushing
any of the EDIT- buttons or the SELECT button one can adjust
displayed parameters.
CONTROL switches between parameter selection and
parameter adjustment mode, selected
push
CONTROL change of parameter selection or
adjustment of selected parameter value
(see menu explanation)
turn
parameter or value is highlighted by arrows
on display
Each time SELECT button is pushed it opens next utility menu.
If a function menu is opened (after pushing related EDIT
button) the SELECT button changes the channel selection.
After finishing of settings ESC button switches back to main
level display. All settings are stored as current adjustment
automatically.
Gain display shows gain setting for all channels. You can jump
to the gain menu from any other edit menu by pushing ESC
button.
Adjustments are made by turning&pushing CONTROL knob as
described previously (see 5.1).
GAIN 1: 0.0 3: 0.0
>M:< 2: 0.0 4: 0.0
M: master control, ganging level settings for all
channels following channel 1
GAIN 1…4: channel independent -15.0 ... +15.0 dB
Mode menu shows sync setting, input and link setting of the
unit. There are two windows available by pushing MODE EDIT
button once or twice.
Adjustments are made by pushing and turning CONTROL knob
(see 5.1). Return to level display with EXIT.
SYNC MODE: selection of sync signal input
CH 1/2 - sync on digital input 1/2
EXT - sync on external sync input
VIDEO - sync on video sync input
SDI - sync on SDI input (only if SDI input
or split mode is selected, no sync
LED lits!)
LINK MODE: all channels independent or following link
combinations:
1+2, 3+4, 1+2 & 3+4, 1+2+3+4
SPLIT AES/EBU + SDI input in split mode (see 4.4)
2. menu
IN12< TR12 IN34 TR34
AES off SDI on
INxx: selection of signal input
AES digital input AES/EBU
SDI SDI input (embedded audio)
TRxx: selection of transparent input
on/off for bit transparent path between
auto input and output (for Dolby E)
If set to AUTO the path is switched to
transparent automatically if the
non-audio flag in the AES/EBU
signal is set.
3. menu (just if SDI interface is present)
SDI GROUPS: > IN< OUT
1 1
IN: selection of SDI group for deembedding
input signals
1...4
OUT: selection of SDI group for embedding
output signals
Expander menu shows expander settings for selected channel.
Adjustments are made by pushing and turning CONTROL knob
(see 5.1). Return to level display with EXIT.
PR CH >EXP< THRS REL 01 2 ON -32 SLOW
PR: number of current preset
CH: selected channel (change with SELECT)
EXP: expander on/off
THRS: threshold level -50 ... -20 dBFS
REL: release time slow, mid, fast
De-esser menu shows de-esser settings for selected channel .
Adjustments are made by pushing and turning CONTROL knob
(see 5.1). Return to level display with EXIT.
PR CH >DEES< TYPE RNG 01 2 ON MALE -8.0
PR: number of current preset
CH: selected channel (change with SELECT)
DEES: de-esser on/off
TYPE: de-essing characteristic male / female
RNG: range of s-frequency reduction -20 ... 0 dB
Compressor menu shows compressor settings for selected
channel .
Adjustments are made by pushing and turning CONTROL knob
(see 5.1). Return to level display with EXIT.
PR CH >CMP< RATIO RNG 01 2 ON 1.5 6
PR: number of current preset
CH: selected channel (change with SELECT)
CMP: compressor on/off
RATIO: compressor ratio 1.0:1 ...4.0:1
RNG: compression range (maximum compression
gain) 0 ... +15 dB
Limiter menu shows limiter settings for selected channel .
Adjustments are made by pushing and turning CONTROL knob
(see 5.1). Return to level display with EXIT.
PR CH >LIM< THRS PRO 01 2 ON -9.0 1
PR: number of current preset
CH: selected channel (change with SELECT)
LIM: limiter on/off
THRS: limiter threshold level -20 ... 0 dBFS
PRO: selected program-preset for adaptive controlled
algorithms
The selection of the parameter PRO in the limiter edit menu changes the
range of time constant values as follows:
PRO adaptive 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 basic Multi-Loop principle of Jünger Audio dynamics processors
operates with adaption of dynamic range control parameters to the
incoming audio signal. That means permanently analysis and calculation
of attack times, release times , thresholds and interaction parameters of
several frequency linear control circuits.
(please refer to chapter 2 also)
Changing of PRO defines 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.
PROCESSOR
C: x.x display of loaded controller software version
D: x.x display of loaded dsp software version
display error / message remedies
NO SYNC no sync at sync input! connect the sync input
NO SDI! SDI input selected, no
Should have remained the device no more operable and/or in the
program execution stand, recommends itself an initialization the
device.
During initialization, all storage areas important for the program
and registers are loaded with the factory setup and the program
is restarted.
Any button is to be held pressed in order to initialize the device
during switch-on of the device until the program started. To the
start of the program and at the completion of the displays (how
described in 7.1), the device is ready for operation with the
factory setup.
After an initialization of the device, all user presets and
adjustments are erased and/or overwritten by the factory
setup!
display of model
valid SDI signal
received!
(selectable in SYNC field) with
valid input signal
¾ CH 1/2: sync on DIGITAL IN
CH 1/2
¾ EXT: sync on SYNC
AES/EBU
¾ VIDEO: sync on SYNC
VIDEO
¾ SDI: sync on SDI input
check the availability of SDI
In digital video recording technology four digital audio channels
are the standard configuration. This channel capacity is used
increasingly in production and post-production for surround sound,
providing mix options and for multi-lingual productions.
Quite often it is necessary to make corrections or changes to the
audio which until now required the use of an expensive digital
audio mixer. These tasks can now be easily solved with the
Jünger Audio range of digital audio toolboxes. Simple processing
for up to four digital audio signals may be carried out quickly and
efficiently.
Using the SDI versions (SDI=Serial Digital Interface, digital
component video format with 270Mb/s transmission) b40 series
can process embedded audio.
The standard allows up to four groups each of four mono audio
channels. Usually used by most of D-VTR's and other equipment
is Group 1 with 48 kHz synchronous sampling. Synchronous
means that the audio clock is genlocked to the associated video.
Each channel can have up to 20 bits of resolution per audio
sample.
The 4-channel processors of b40 series fitted with SDI-interface
are compatibel with the standard SMPTE 272M-AB. They support
48 kHz synchronous audio sampling with 20 bit word length.
The Jünger Audio SDI interface provides for one group of four
audio channels to be extracted from or inserted into the SDI data
stream. To address a specific channel group the group selection is
possible (see 4).
The b40 provides an optional SD- or HD-SDI board. When you
switch on the device the plugged in interface will be indicated in
the display
FEATURES
• Bypass relay for SDI IN >SDI OUT
• Bit transparent for coded data streams (e.g. DOLBYE/20bit)
• De-embedder: user selectable de-embedding of one group
• Embedder: user selectable embedding to one of 4 groups
• SDI-SYNC: SDI input can be the clock source of the device
• For HD-SDI: Multi-Format HD/SD operation with auto detection
For the basic working mode the input of the digital audio processing
can be selected between AES/EBU or SDI (serial digital video with
embedded audio). The processed signals are present at both outputs
always - at AES/EBU and SDI.
There are two additional working modes using the SDI interface. SDI
Bypass is bypassing the SDI data stream. In this case only extracted
audio is processed and available at AES output. In Split Mode the
audio path is splitted. Embedded audio can be processed with external
equipment via AES interface.
Following illustration shows working modes:
data rate : 270 Mb/s, 525/625 Line rate
format : serial digital component video 4:2:2
with embedded audio
(ITU-R BT.601, SMPTE 272M-A)
level: 800 mV +/- 10%
equalisation : appr. 180 m (Belden 8281)
audio data: 4 channels, 20 bit
features : SDI relais bypass
power supply : +5V DC
consumption : approx. 500 mA
dimension : 3RU, 4HP, 160mm depth (EUROPA size pcb)
temperature : 10°C to 40°C
humidity : 90%, non condensing
supported video standards:
HD 720/60 SMPTE 296M HD 1080/25 SMPTE 274M
HD 720/50 SMPTE 296M HD 1080/24 SMPTE 274M
HD 720/30 SMPTE 296M HD 1080/50 SMPTE 295M
HD 720/25 SMPTE 296M HD 1035/60 SMPTE 260M
HD 720/24 SMPTE 296M
HD 1080/60 SMPTE 274M SD 525/59.94 SMPTE 125M
HD 1080/50 SMPTE 274M SD 625/50 SMPTE 125M
HD 1080/30 SMPTE 274M
all HD-standards are supported also with their 1/1001-frame-rates
AUDIO :
audio data format : 24 Bit, transparent for C-Bit and U-Bit according to
AES3
audio sample rate : 48 kHz synchronous to video-carrier (SD and HD)
32 kHz ... 48 kHz asynchronous to video-carrier (HD
only)
latency : (deembedder + embedder)
HD : < 800µsec
SD : < 2,6 msec
GENERAL :
power supply : +5V DC
consumption : approx. 1.000 mA
dimension : 3RU, 4HP, 160mm depth (EUROPA size pcb)
temperature : 10°C to 40°C
humidity: 90%, non condensing