ADM20 THE REFERENCE CREATED BY DIGITAL PRECISION
The ADM20 is an active, two-way studio monitor for medium to high volume levels. The signal
processing, including equalization, crossover and correction of the transient behaviour of the
components, is completely done at the digital level. A connection to an existing, digital studio
environment can be made digitally using theAES3 interface. An analogue feed is however, also
possible; a high quality 27 bit gain staging, sigma-delta Converter with 64 x oversampling and a
dynamic of more than 130 dB is responsible for the conversion. The digital signal is further
processed bya fast 60MHz floatingpoint SharcDSP in our proprietarydesigned processor board.
This convertsthe 24 bitwideinput datainto a floatword with 32bit mantissaand 8Bit exponentthat
theoretically allows a dynamicof over 1500 dB; noise from overloading, scaling and rounding off
are therefore eliminated.Apatented system (Pat. Nr: 198 23 110) raisessignals with a word width
of less than 24 bits to an audiophile level. Parameters that can be externally adjusted, such as
volume, filter etc., work directly at the floating point level in the process so that also at this stage,
e.g. when adjusting the volume, no scaling errors occur. On the output side, the DSP works with
two 24 bit D/A converters feeding the high performance MOFSET output stage, that is equipped
with individually selected transistors. With a bandwidth of 100kHz and a rise time of 80 V/us, the
output stageperforms to 100W in treble range and 200Wat deep mid-range. The resulting sound
pressure is 116dB cont. and 122dB peak.
SPECIFIC FEATURES OF DIGITAL SIGNAL PROCESSING
Digital signal processing must meet the following requirements:
- Separation of the relevant signal parts into treble and bass/ mid-range as well as the 3rd way on
the analogueoutput socket
- Equalization of the transmission behaviour of each component
- Overloadprotection function
Everyanalogue interference with thesignal, - e.g. insertion of a cross-over filteror equalization of
individual speakers - causesnot onlythedesired changebutalso a change totheimpulse behaviour
of the system.Analogue equalizationwould inevitably lead to a shortimpulse beingsmudged along
the time axis and create a later oscillation of the impulse response at the end of the equalization.
Additionally,an impulse would,in turn,cause afurther oscillationdue to theinertiaof themembrane.
Putting these together cause a deformation of a right-angled input signal to a less related output
signal characteristic. The FIRTEC ™ equalizer (Finite Impulse Response) which was developed
for and is used in the ADM20, is free of those distortions described above.
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