54-16
54-4
60849
Pace Optimisation for Maximum Intelligibility
The Speech Temporal Transposition Algorithm automatically adjusts the pace of voice announcements to maximise their intelligibility in any acoustic environment. The system analyses the type of voice (male or female), determines the speech tempo, and — most critically — detects and shortens excessive prolongation of individual phonemes (so-called "stutters").
The result is that announcements are naturally and evenly slowed down or sped up as required, while voice timbre and naturalness are fully preserved. This significantly increases intelligibility in high-reverberation environments — railway stations, airports, underground garages, stairwells, and sporting venues — where a fast-paced message would otherwise be lost to acoustic reflections.
Automatically adjusts speech pace without operator intervention
Detects and corrects over-articulation of individual phonemes
Preserves natural voice timbre throughout the adjustment
Adapts pace to the STI index measured in each individual zone
Works on both pre-recorded messages and live operator announcements
Real-Time Sound Adaptation to Acoustic Conditions
The Adaptive Filtration Algorithm provides automatic adjustment of volume and sound timbre based on continuous real-time readings from the SMART-ANSM-01 noise sensing microphones. The system analyses the noise spectrum and reacts to changes by applying narrowband filters, ensuring announcements remain intelligible regardless of the ambient noise environment.
The algorithm maintains a consistent headroom (5, 10, or 15 dB selectable) between the announcement signal and any acoustic background noise, automatically correcting the frequency response of each zone in real time. The STI coefficient is constantly maximised across the entire installation — with no manual intervention required.
Continuously adjusts SPL, timbre, and frequency response per zone
Analyses noise spectrum in real time and applies narrowband filters
Maintains selectable headroom (5, 10, or 15 dB) over background noise
Responds instantly to environmental changes — e.g. ventilation, crowd noise
Works zone by zone — each area optimised independently
SMART-ANSM-01 microphones measure background noise level, frequency characteristics, and reverberation time (RT) in each zone.
The SMART-DU1604 processor calculates SNR, STI, and RT parameters for each zone from the incoming microphone data.
The AF algorithm automatically adjusts SPL, timbre, and frequency characteristics per zone to maintain target speech intelligibility.
The STTA algorithm analyses and adjusts the speech pace of each announcement, slowing or speeding it to match the STI in each zone.
Continuous monitoring means the system automatically re-optimises if acoustic conditions change — fully hands-free, in real time.
SMART-CU-8LCD
Stand-alone Control Unit with 8 control slots and touch screen GUI
A matrix mixer of input signals routing to four 100V internal audio buses, a 48-channel digital system bus, or directly to audio outputs. Equipped with dual redundant processor card (2× ABT-xCPU) and a 4.5″ LCD colour touch display for direct system management and monitoring.
| Power Supply | 48V DC (40–57V DC) |
| Power Consumption | Up to 60W |
| Control Slots | 8 |
| Function Slots | 4 |
| Display | 4.5" LCD touch screen |
| Simultaneous Messages | 8 in different zones |
| SD Memory | Up to 32 GB (48 kHz / 16-bit) |
| Network Ports | 2 × 1Gbit + 1 × POE |
| Dimensions | 482 × 85 × 325 mm |
| Weight | Up to 8 kg |
| Mounting | 19" rack mount |