DriveSYS8-ae 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 AES67 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-ae 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 AES67 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
