254
Max Network Units
660
Speaker Lines
180
Simultaneous Messages
SMART-CU-8LCD
EN
54-16
Voice Alarm Control & Indicating Equipment
EN
54-4
Power Supply Equipment
EN
60849
Sound Systems for Emergency Purposes
Cert No: 1438-CPR-0948
STTA
Speech Temporal Transposition Algorithm

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.

Key Benefits

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

AF
Adaptive Filtration Algorithm

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.

Key Benefits

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

01
Acoustic Recording

SMART-ANSM-01 microphones measure background noise level, frequency characteristics, and reverberation time (RT) in each zone.

01
Data Analysis

The SMART-DU1604 processor calculates SNR, STI, and RT parameters for each zone from the incoming microphone data.

01
Adaptive Filtration

The AF algorithm automatically adjusts SPL, timbre, and frequency characteristics per zone to maintain target speech intelligibility.

01
Speech Transposition

The STTA algorithm analyses and adjusts the speech pace of each announcement, slowing or speeding it to match the STI in each zone.

01
Dynamic Adjustment

Continuous monitoring means the system automatically re-optimises if acoustic conditions change — fully hands-free, in real time.

EN 54-16 · 1438-CPR-0948

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