Public Address and Voice Evacuation System
The Smart approach to better understanding
smartVES is at the stage of pre-implementation works and during the certification process – it will soon be available on the market.
SmartVES is a new PAVA system utilising innovative technologies that improve the intelligibility of broadcast voice messages in buildings with difficult acoustic conditions.
SMART-DU-1604
smartVES is a unique solution in that, regardless of the prevailing acoustic conditions in the facility, smartVES maximizes the speech intelligibility parameter dynamically, using the full technical capabilities of speakers, amplifiers and the settings of signal processors. The advanced method of auto-calibration and of mapping the matrix of SMART – ANSM-01 measuring microphones to the loudspeaker zones makes the complete system extremely simple and quick to configure with most of the settings being selected automatically.
Tha advantages of smartVES technology:
- simplicity of configuration;
- quick start-up, thanks to the auto-calibration function, the system detects the positions of the SMART-ANSM-01 measuring microphones in relation to the loudspeaker zones;
SMART-xCtrLine-44
The smartVES system uses the new SMART-xCtrLine-44 loudspeaker line control cards, which allow full freedom of routing independent voice messages from the system BUS audio buses, to each of the 4 outputs of the card.
SMART-ANSM-01
The measurement microphones are an integral part of the smartVES system. Their purpose is to test the level and frequency characteristics of sound at an object in the range of 100Hz-10kHz (+/- 2dB) with sound levels from 50dB to 120 dB SPL.
SMART-AMAP-6
The SMART-AMAP-6 is a measurement microphone aggregation point that receives the signal from 6 measurement microphones and enables the addressing and use of their return signals in the adaptive filtering algorithms of the assigned zones.
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.
