Junger Audio d02 User Manual

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jungeraudio

d02

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

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CONTENTS

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

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. 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 ............................................

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

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

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 .......................................................................

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

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

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0

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.

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.

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.

Basic
Functions

1.2.

The
Jünger Audio
Dynamics
Processor
Principle

1-1

1. THE DESIGN OF THE DEVICE

traditional compressor
and limiter designs

multi-band structure

multi-loop principle

delay time

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).

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1. THE DESIGN OF THE DEVICE

f

p

A A

Multi - Band

f

delay

1

2

n

Multi - Loo

1

fig. 1:
basic principles of

dynamic range
processors

2

m

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
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, air­conditioning, hum and noise).

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

compressor

compression gain

expander

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1. THE DESIGN OF THE DEVICE

)

)

g

)

)

fig.. 2:
static
characteristics:
compressor

static characteristics: co mpressor

output level d02 (dBFS

0

2.0 : 1

1.6 : 1

-10

1.3 : 1

-20

1.1 : 1

-30

fig. 3:
static
characteristics:
compressor

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-50

-50 -40 -30 -20 -10 0

compression gain: max. 10 dB
parameter: ratio

static characteristics: compressor

output level d02 (dBFS

0

-10

-20

15 dB

off

input level (dBFS

1-4

-30

-40

10 dB

5 dB

off

-50

-50 -40 -30 -20 -10 0

input level (dBFS

compression gain: max. 15 dB
parameter: compression
ratio: 1.6 : 1

ain

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.
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.

1.3.

A/D­Conversion with
Digital Full Scale
Level

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