KS DIGITAL ADM20, Руководство пользователя

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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 the AES3 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 by a fast 60 MHz floating point Sharc DSP in our proprietary designed processor board.
This converts the 24 bit wide input data into a float word with 32 bit mantissa and 8 Bit exponent that
theoretically allows a dynamic of over 1500 dB; noise from overloading, scaling and rounding off
are therefore eliminated. A patented system (Pat. Nr: 198 23 110) raises signals 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 stage performs to 100W in treble range and 200W at 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 3 rd way on
the analogue output socket
- Equalization of the transmission behaviour of each component
- Overload protection function
Every analogue interference with the signal, - e.g. insertion of a cross-over filter or equalization of
individual speakers - causes not only the desired change but also a change to the impulse behaviour
of the system. Analogue equalization would inevitably lead to a short impulse being smudged 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 a further oscillation due to the inertia of the membrane.
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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