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© 2018 by TVV Sound / Proudly created by Schiettecatte J

The VX Engine, a fan-free 2RU rack-mount device with enormous processing power, is the heart of the system. It provides all the call control and audio processing needed for the system, and supports up to 30 active calls on-air simultaneously, across as many as 20 studios. Its two Gigabit Ethernet ports provide a cost-effective interface to both telephone lines and studio audio via proven Livewire AoIP. VX is Web-based, so remote control and configuration are a snap — engineers can work with it from any place they can get online.

Call processing is sophisticated and flexible. Lines may be readily shared among studios; the Web interface allows easy assignment of lines to “shows”, which can then be selected by users on the studio controllers. Each studio can provide its own Program-on-Hold audio to callers.

Audio processing features also have taken a leap forward. The processing power of the VX Engine allows multiple calls to be conferenced and aired simultaneously, with excellent quality. The hybrids are equipped with a rich toolbox to make caller audio sound its best, no matter what kind of line or phone the caller uses. Caller audio benefits from Smart AGC coupled with famous Telos three-band adaptive Digital Dynamic EQ and a three-band adaptive spectral processor. Send audio gets its own sweetening with a frequency shifter, AGC/limiter, and FhG’s Advanced Echo Cancellation technology that literally eliminates open-mic feedback. Call ducking and host override are part of the VX toolkit as well, and
talent can manage and customize their telephone settings and workflow using VX Show Profiles to store and recall commonly used show configurations.

You’ll notice that there are no audio I/O or telco ports on the VX Engine. All connections to the Engine are via the two Ethernet jacks that connect to your system’s Ethernet switch to support a wide variety of peripherals: telephone lines, Livewire studio audio, VSet phones, VX Producer PC applications, console- integrated controllers, etc.

For traditional phone services, you can choose standard telco gateways from Asterisk, Patton, Cisco, Grandstream, and others to connect to T1/E1, ISDN, and POTS providers. And, if you have a VoIP-based PBX or SIP Trunking telco service, the VX uses standard SIP (Session Initiation Protocol) and RTP (Real- time Transport Protocol).

 

General

  • Telos 5th-generation Adaptive Digital Hybrids.

  • Maximum number of phone lines: 48, when used with a-Law or u-Law codecs for VoIP lines. (Higher quality codecs, such as G.722, consume more system resources and result in a decreased number of total lines available.)

  • Maximum number of SIP numbers: 250

  • Maximum active on-air calls: 48

  • Maximum number of simultaneous audio connections (Livewire+ I/O channels): 16 systemwide

  • Maximum on-air calls on one fader: 4

 

Analog Inputs (with Telos Alliance xNode):

  • Input Impedance: >40 k Ohms, balanced

  • Nominal Level Range: Selectable, +4 dBu or -10dBv

  • Input Headroom: 20 dB above nominal input

 

Analog Outputs (with Telos Alliance xNode):

  • Output Source Impedance: <50 Ohms balanced

  • Output Load Impedance: 600 Ohms, minimum

  • Nominal Output Level: +4 dBu

  • Maximum Output Level: +24 dBu

 

Digital Audio Inputs And Outputs

  • Reference Level: +4 dBu (-20 dB FSD)

  • Impedance: 110 Ohm, balanced (XLR)

  • Signal Format: AES-3 (AES/EBU)

  • AES-3 Input Compliance: 24-bit with selectable sample rate conversion, 32 kHz to 96kHz input sample rate capable.

  • AES-3 Output Compliance: 24-bit Digital Reference: Internal (network timebase) or external reference 48 kHz, +/- 2 ppm

  • Internal Sampling Rate: 48 kHz

  • Output Sample Rate: 44.1 kHz or 48 kHz

  • A/D Conversions: 24-bit, Delta-Sigma, 256x oversampling

  • D/A Conversions: 24-bit, Delta-Sigma, 256x oversampling

  • Latency <3 ms, mic in to monitor out, including network and processor loop

 

Frequency Response

  • Any input to any output: +0.5 / -0.5 dB, 20 Hz to 20 kHz

 

Dynamic Range

  • Analog Input to Analog Output: 102 dB referenced to 0 dBFS, 105 dB “A” weighted to 0 dBFS

  • Analog Input to Digital Output: 105 dB referenced to 0 dBFS

  • Digital Input to Analog Output: 103 dB referenced to 0 dBFS, 106 dB “A” weighted

  • Digital Input to Digital Output: 138 dB

 

Total Harmonic Distortion + Noise

  • Analog Input to Analog Output: <0.008%, 1 kHz, +18 dBu input, +18 dBu output

  • Digital Input to Digital Output: <0.0003%, 1 kHz, -20 dBFS

  • Digital Input to Analog Output: <0.005%, 1 kHz, -6 dBFS input, +18 dBu output

 


Crosstalk Isolation, Stereo Separation and CMRR

  • Analog Line channel to channel isolation: 90 dB isolation minimum, 20 Hz to 20 kHz

  • Analog Line Stereo separation: 85 dB isolation minimum, 20Hz to 20 kHz

  • Analog Line Input CMRR: >60 dB, 20 Hz to 20 kHz

 

VX engine 

IP/Ethernet Connections

  • One 100BaseT/gigabit Ethernet via RJ-45 LAN connection

  • One 100BaseT/gigabit Ethernet via RJ-45 WAN connection

 

Processing Functions

  • All processing is performed at 32-bit floating-point resolution.

  • Send AGC/limiter

  • Send filter

  • Gated Receive AGC

  • Receive filter

  • Receive dynamic EQ

  • Ducker

  • Sample rate converter

  • Line Echo Canceller (hybrid)

  • Acoustic Echo Canceller (wideband)

 

Power Supply AC Input

  • Modular, field-replacable auto-sensing supply, 90VAC to 240VAC, 50 Hz to 60 Hz, IEC receptacle, internal fuse

  • Power consumption: 100 Watts

 

Operating Temperatures

  • -10 degree C to +40 degree C, <90% humidity, no condensation

 

Studio Audio Connections

  • Via Livewire+ IP/Ethernet. Each selectable group and fixed line has a send and receive input/output.

  • Each studio has a Program-on-Hold input.

  • Each Acoustic Echo Canceller has two inputs (signal and reference) and one output.

  • Livewire+ equipped studios may take the audio directly from the network. Telos Alliance xNodes are available for pro analog and AES3 breakout.

 

Telco Connections

  • Audio: standard RTP. Codecs: g.711u-Law and A-Law, and g.722

  • Control: standard SIP trunking

VX Engine - VX Broadcast VoIP

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