"The Room" is an advanced 8-channel DSP with amazing specifications and skills. It offers you an untouchable versatility and a really easy-to-use interface, developed for customers with the highest ambitions. A unique design, with six seperated shielded "tubes". - Each module (ADC, DAC, DSP) has its own tube for maximum decoupling from each other.
Front and back pannel milled from solid 30mm aluminium.

- High-end audio signal processing system

- DSP with 32/40-bit, 192kHz system sample rate

- FIR filters with linear or minimal phase

- High-quality signal converters

- Flexible IO components

- Ultimate materials and housing design

- Unique free measurement and filter design program 

The analog digital converter (ADC)

This module transfers the best analog signal quality losslessly to the digital level.

- Completely symmetrical construction with dual mono operation of the AD converters (> 24 bits, 192kHz sample rate)
- Four balanced inputs, 2 x Line and 2 x Microphone or Phono,
- Two ultra high-quality analog preamplifiers, optionally with Lars Lundal transformer, impedance matching, and 48V phantom power
- Extremely elaborate, separate power supply, only a negligible residual current flows through the ground
- Use of the best operational amplifiers, capacitors, and non-magnetic resistors
- Galvanic isolation of the module from the digital part (DSP) by fastest data couplers

The digital analog converter (DAC)

This converter generates the best analogue signal quality from the digital level.

- Completely symmetrical construction with dual mono or single operation of the DA converters (external DSP oversampling filter)
- Two balanced or four unbalanced outputs
- Very high-quality, low-noise current-voltage conversion
- Extremely elaborate, separate power supply, only a negligible residual current flows through the ground
- Analogue volume control, mute, -96 to 0dB in 0.5dB steps
- Relay mute without switch contact in the signal path
- Use of the best OP amps, capacitors, and non-magnetic resistors
- Galvanic isolation of the module from the digital part (DSP) by fastest data couplers

Signal processing/measurement program and filter design

The combination of hardware and software enables the optimal tuning of all loudspeaker systems.

Analog Devices Signal Processor

- Sharc signal processor with 400MHz 32/40-bit floating-point arithmetic
- Master version with USB audio (32Bit/192kHz, max. 8 channels)
- SDRAM for adapted runtimes
- Digital IO: SP/DIF, AES/EBU, I2S, TDM 
- Function and filter code:
Setup and control ADC/DAC
Activation of inputs and outputs
RS232, I2C, displays, buttons
FIR and IIR filters
Special oversampling

Measure and evaluate

- Completely proprietary development
- Measurements with various test signals
ATB, MLS, sinus sweep, noise...
- Distance/time measurement
- Averaging and rounding
- Evaluation/presentation:
Frequency response, oscillogram, phase, delay, waterfall, RT60, RMS, IMD, THD, 3D...
- Tools:
Signal generator, oscilloscope, spectrum

Filtering and equalizing

- All analog standard filters like
Biquads, Butterworth, Chebyshev, Linkwitz-Riley
- Freely definable FIR filter curves:
FFT filter, inverse on range
- Implementation of filter curves to FIR filter with various phase responses
(original, minimal, linear)
- IIR filter/equalizer designer with touch pen to "draw the Curve"
- Uploading filters to DSP
- Delay management for all channels

Housing in tube design

Unique design, with three or six "tubes". Separate power supply.

- Front and back (30mm) milled from solid aluminium
- Adjustable arrangement of connections on the back (IO panels)
- Operation via two rotary encoders (with precision bearings) or touch-colour display
- Three-point mounting, IR receiver integrated in the front foot
- Each module (ADC, DAC, DSP) has its tube for maximum decoupling from each other.
- Separate power supply for optimal separation of network and usage voltage.