DriveSYS8-Dante is an 8 × 8 audio processor with advanced auto-mixing, matrix, full DSP and multi-zone management, designed for commercial, corporate and conference installations. It features 8 balanced analog inputs and 8 balanced analog outputs, with Dante support for 32 inputs and 32 network outputs. Each input features microphone preamplification with 48 V Phantom power, adjustable gain from 0 to 40 dB in 1 dB steps, and 24-bit AD/DA conversion, ensuring high audio quality and maximum configuration flexibility.
The DSP integrates professional modules including feedback suppression, automix, matrix mixing, zone routing, PEQ, crossover, compressor, AGC, AEC, and AI ANC, as well as advanced features such as noise gate, phase filter, high-pass and low-pass filtering, limiter, dynamic DEQ, and dedicated loudspeaker libraries.
DriveSYS8-Dante supports remote control via TCP/IP, RS232, RS485, and USB (driver-free), with a comprehensive web interface for configuration, presets, loudspeaker tuning, FIR, DEQ, delay, and user management. The front panel includes an LCD display for device name, presets, IP, input/output levels, input mode, and firmware version.
User benefits
- High audio quality thanks to a comprehensive DSP with advanced modules
- Precise input, output, and zone management for multi-room installations
- Intelligent automixing for multi-microphone scenarios
- Network integration via Dante for distributed systems
- Flexible remote control via TCP/IP, RS232, RS485, and USB
- Intuitive web interface for quick and complete configuration
- Scene and speaker presets for instant configuration recall
- Dedicated processing for each channel with PEQ, DEQ, FIR, delay, and limiter
- Front-panel LCD display for instant parameter monitoring
- Security features with predefined lock and mask settings for project protection
Areas of use
- Conference rooms and meeting rooms
- Auditoriums and theaters
- Corporate spaces and executive offices
- Commercial multi-zone systems
- Stadiums, arenas, and sports facilities
- Multi-purpose halls and conference centers
- Voice announcement systems in complex buildings
- Government and institutional installations
- Classrooms and environments training
- Hotels, resorts, and hospitality facilities
Technical characteristics
DSP chip
- Signal processing: 32 bit float point DSP 1 GHz
- Audio system latency: 3 ms
- D/A conversion: 24 bit
- Sampling rate: 48 kHz
Analog audio inputs and outputs
- Input channels: 8 balanced analog inputs, MIC/line level
- Audio connectors: 3.81 mm Phoenix connector, 12 pin
- Input impedance: 7 kohm
- Maximum input level: 20 dBu (7.746 VRMS) / line; 0 dBu (0.774 VRMS) / MIC @ 20 dB gain
- Phantom power: +48 VDC, 5.5 mA per channel
- Output channels: 8 balanced analog outputs, line level
- Output impedance: 87 ohm
Audio performance specifications
- Frequency response: 20 Hz – 20 kHz (±0.5 dB) line; 20 Hz – 20 kHz (±0.8 dB) MIC @ 20 dB gain
- Noise floor: 20 Hz – 20 kHz, A‑wt, −93 dBu / 15.5 µV
- THD + N: −90 dB (@ 0 dBu, 1 kHz, A‑wt) line; −76 dB (@ 0 dBu, 1 kHz, A‑wt) MIC @ 20 dB gain
- SNR: 114 dB (@ 20 dBu, 1 kHz, A‑wt) line; 105 dB (@ 0 dBu, 1 kHz, A‑wt) MIC @ 20 dB gain
- Channel isolation: −110 dB (@ 0 dBu, 1 kHz) line; −120 dB (@ 0 dBu, 1 kHz) MIC @ 20 dB gain
Communication ports and indicators
- USB: built‑in USB drive 31 MB
- RS232: serial port communication
- RS485: serial port communication
- TCP/IP network port: RJ‑45
- Indicators: power LED
Electrical
- Power supply: 100 – 240 VDC 50/60 Hz
- Operating temperature: −20 to 60 °C
Physical
- Net weight: 3.6 kg
- Device dimensions (W × D × H): 48.3 × 27.6 × 4.45 cm