smartVES
The Smart approach to better understanding
SCALABLE SMART PAVA SYSTEM
EN54-16 CERTIFIED
SmartVES is an intelligent PAVA system designed to enhance building security by maximising the intelligibility of voice messages in acoustically demanding facilities.
The system’s architecture is based on digital signal distribution over an IP network, allowing the flexibility to create centralised and distributed topologies.
smartVES is designed according to the NOSPOF (NO SINGLE POINT OF FAILURE) concept, eliminating single points of failure. In its basic configuration, the system has no master unit, the failure of which would cause a complete system shutdown. For demanding applications, full redundancy of all system components is possible.
The unique power distribution concept of the multi-channel amplifiers allows a choice between assigning an amplifier channel to a loudspeaker zone (direct connection) or to a group of zones (BUS topology), taking into account a spare amplifier and an intelligent algorithm to protect the system from amplifier channel overload. In this way, the power of the available amplifiers can be safely and cost-optimally utilised, while meeting the requirements of message allocation to loudspeaker lines. In addition, this reduces power consumption in STAND-BY mode.
The modular design of the system components, including the control units, allows optimum adaptation to the developer’s needs, meeting all design guidelines.
Thanks to the extensive interface, extensive integration with PSIM, BMS or PIS (Passenger Information System) systems is possible, which is particularly important in large and complex projects such as airports, subways or industrial applications.
All system functions have been designed and certified in accordance with EN 54-16.
The adaptive filtration (AF) algorithm
smartVES ensures the maximization of the STI coefficient and maintains the appropriate headroom (5, 10 or 15 dB) between the useful signals, e.g. spoken messages, and any undesirable acoustic background.
This process of the maximisation of the STI is calculated by a number of innovative algorithms implemented in the SMART-DU-1604 unit. The key algorithms include the algorithm for adaptive filtering and temporal transposition of the speech signal, as well as algorithms for calculating SNR, STI and auto-calibration of the system, supporting real-time measurements, whose task is to obtain information that allows optimal adjustment of the sound to the current acoustic conditions.
The speech temporal transposition algorithm (STTA)
The speech temporal transposition algorithm (STTA) naturally and evenly changes the pace and duration of messages spoken in real time through smartVES system microphones (DFMS, DMS, DMS-LCD). The algorithm can distinguish the type of voice (male / female), determines the rate of speech, and most importantly, detects and shortens the duration of stuttering, while stuttering is understood as excessive prolongation of the articulation of the selected sound.
Components of the smartVES system
Adaptive audio processor
SMART-DU-1604
Regardless of the acoustic conditions in the facility, smartVES uses digital sound processing tools to dynamically maximize speech intelligibility. This is achieved through a series of algorithms implemented in the SMART-DU-1604 unit, primarily including the adaptive filtering algorithm and the time-domain speech signal transposition. Additionally, SMART-DU-1604 is equipped with real-time algorithms responsible for calculating parameters such as signal-to-noise ratio (SNR), speech transmission index (STI), and reverberation time (RT).
Control Units
SMART-CU-8/11
The SMART-CU-8/11 Control Unit is a modular device that offers exceptional capabilities for expanding the number of processed audio signals both on input and output. With 8 or 11 control slots for dedicated cards and 4 functional slots intended for connecting sources such as CD, radio, or advertising server, SMART-CU-8/11 provides remarkable flexibility and extensive configuration possibilities. Maximizing the use of DSP resources allows the unit to handle 16 analog audio input signals, 32 output signals, and 8 digital audio signals in the smartVES network. Additionally, each ABTxAudIO4/8RS card offers advanced functions such as a 3-band parametric equalizer on audio inputs, an 8-band parametric equalizer, a delay line, a feedback eliminator, and a limiter on audio outputs.
Adaptive Noise Sensing Microphone
SMART-ANSM-01
Measurement microphones are an integral part of the smartVES system. Their purpose is to test the sound level and frequency characteristics in the range of 100 Hz to 10 kHz (±2 dB) at sound levels from 50 dB to 120 dB SPL.
Adaptive Microphone Aggregation Point
SMART-AMAP-6
The SMART-AMAP-6 interface allows the connection of up to 6 monitored measurement microphones, which will be used by adaptive filtering algorithms in the assigned zones.
Power Amplifiers
SMART-PAXXXX/E
The SMART-PAXXXX/E is a family of 1/2/4/8-channel Class D power amplifiers designed for mounting in a 2U rack, intended for distributed 100 V speaker systems. The SMART-PA8080/E amplifier can deliver up to 8 × 80 W, while the SMART-PA8160/E and SMART-PA2650/E models increase power to 8 × 160 W and 2 × 650 W respectively. In bridged mode, the amplifier channels are combined and can deliver 4 × 160 W for SMART-PA8080/E, 4 × 320 W for SMART-PA8160/E, and 1 × 1300 W for SMART-PA2650/E. These amplifiers feature a 48 VDC input, allowing connection to a battery backup system to ensure maximum availability and durability in an audio warning system.
Power Supply Equipment
SMART-PSM48
The SMART-PSM48 power manager is designed to distribute direct current (DC) power from power supplies (SMART-PS48800) and a backup battery. The device controls battery charging and distributes power to all devices in the Voice Evacuation System (VES) with a maximum current of 60 A. When the system uses battery backup, the delivered power is 3.2 kW (48 V). In standby mode, it uses a battery pack with a capacity of up to 200 Ah as a power source.
Fireman’s Microphone
ABT-DFMS
The smartVES ABT-DFMS firefighter microphone is a monitored external device that works with control units in a redundant communication ring. This allows it to also serve as the master control unit of the system. The firefighter microphone can be used to activate alarm messages, make general public announcements, select individual zones, and broadcast live voice messages. It is equipped with programmable function buttons that can be assigned functions as required. Up to 5 ABT-EKB-20M keyboard extensions with additional function keys can be connected.
Zone Microphone
ABT-DMS
This zone microphone is used to activate general public announcements, select individual zones, and broadcast live voice messages. It can be connected directly to the selected control unit or via an additional Ethernet switch. The zone microphone can be powered locally (48 V) or from the control unit via POE. The microphone is equipped with programmable function buttons that can be assigned functions as required. All operational parameters can be programmed, such as assigning zones to different keys, naming zones and zone groups, setting priorities, configuring access rights to announcements, adjusting volume, the “push to talk” button, as well as routing and toggling music on and off.
Zone Microphone with LCD
ABT-DMS-LCD
This zone microphone performs the same function as the ABT-DMS zone microphone. To make its operation easier and more intuitive, the microphone is equipped with an LCD touchscreen. The LCD display allows for clear visual feedback and straightforward interaction, enabling users to quickly and efficiently select zones, adjust settings, and broadcast announcements. This enhanced interface improves usability, ensuring that even complex configurations can be managed with ease.