COVER COVER
COVER
Issue 7 / April 2017
COVER
TABLE OF CONTENTS | TELOSALLIANCE.COM
Issue 7 / April 2017
Telos®
Audio Codecs & Transceivers
Axia®
Networked Radio Consoles
Z/IP ONE
Fusion
Zephyr Xstream
Element
Zephyr iPort PLUS
SoftSurface
Broadcast Telephone Systems
iQ
VX Prime
Radius
VX Broadcast VoIP
RAQ
Hx6 6-Line Talkshow System
DESQ
iQ6 6-Line Talkshow System for Livewire
StudioEngine Mixing Engine
Hx1 & Hx2 Digital POTS Hybrids
PowerStation Mixing Engine
Omnia®
Radio Processing
Omnia.11
QOR.32 Mixing Engine
QOR.16 Mixing Engine
IP Audio Network Routing & Control
Omnia.9
xNodes
Omnia.9sg
xSwitch
Omnia.7AM
xSelector
Omnia.7FM
Routing Control Panels
Omnia VOLT
Studio Control Panels
Omnia ONE
Axia Livewire+ AES67 IP-Audio Driver
Microphone Processing and Management
Omnia VOCO 8
Pathfinder Routing Control Software
Pathfinder Core PRO Routing Control &
Facility Management Appliance
iProbe Network Management Software
iPlay Networked Stream Player
iProFiler Networked Audio Archiving
Networked Intercom
IP Intercom
TABLE OF CONTENTS | TELOSALLIANCE.COM
Issue 7 / April 2017
25-Seven®
Watermark Monitoring and Enhancement
Linear Acoustic®
TV Loudness Processing
Voltair
AERO.2000
TVC-15
AERO.100
Radio Profanity Delay
Program Delay Manager
Audio Time Management
Precision Delay
AERO.10
AERO.soft
Interface
SDI xNode
Audio Quality & Loudness Monitoring
Z/IPStream
Streaming Audio Processing + Encoding
LQ-1000
MT2000
Z/IPStream X/2 Streaming Software
Z/IPStream 9X/2 Streaming Software
Z/IPStream A/XE Streaming Software
Minnetonka
Streaming Audio Processing + Encoding
Z/IPStream F/XE File Encoding Software
AudioTools Server
Z/IPStream R/1
Audio Tools Cloud
Z/IPStream R/2
Audio Tools Focus
Audio Tools Carbon
SurCode for Dolby E
SurCode for Pro Logic II
SurCode for Dolby Digital Plus
TELOS | Z/IPONE
Z/IP ONE IP Codec
The IP Codec that drops jaws. Not audio.
OVERVIEW
Z/IP ONE is a 1 RU rack-mount IP codec for remote broadcasting. It’s a single-space rack unit perfect
for studios, TOCs and remote kits. Z/IP ONE Includes a full range of codecs including AAC-ELD, AAC-HE,
AAC-LD, MPEG 4 AAC, MPEG 2 AAC, MPEG Layer 2, G.711, G.722 codecs, plus linear audio and optional
aptX® Enhanced coding. Z/IP ONE supports SIP 2.0 protocol and conforms to N/ACIP standards; it
also works with VoIP devices and connects to compatible SIP PBXs. A full complement of I/O, including
Livewire® AoIP, analog and AES/EBU, is standard.
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TELOS | Z/IPONE
FEATURES
• Works with wired and wireless IP connections including WiFi, WLAN and UMTS/EVDO networks;
includes matching Wi-Fi stick.
• Exclusive Agile Connection Technology (ACT) automatically senses network conditions and adapts
codec performance to provide the best possible audio.
• Largest choice of high-performance codecs: AAC-ELD, AAC-HE, AAC-LD, MPEG Layer-2, MPEG-4 AACLC, MPEG-2 AAC-LC, G.711, G.722, and linear PCM. Enhanced aptX coding optional.
• Dual Ethernet ports for separate streaming and control, LAN for local control with Livewire audio and
GPIO; separate WAN for secure connection to wide area networks.
• Livewire, analog and AES/EBU I/O standard.
• Easy browser setup via built-in Web server.
• “Push Mode” for one-way network transmission.
• “Multiple Push Mode” for audio distribution to multiple destinations.
• Distributed Z/IP Server directory service, with multiple geolocations, lets you easily connect to
other Z/IP ONE devices without the need for an IP address and also provides sophisticated NAT
traversal support.
• Transparent, time-aligned RS-232 channel for remote control or metadata, e.g., RDS.
• Time-aligned 8-bit parallel GPIO port for signaling and control.
• Slim 1RU form factor is perfect for studio racks, remote kits or road cases.
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TELOS | Z/IPONE
IN DEPTH
Z/IP ONE: It’s The Zephyr® for IP
These days, you can get broadband Internet just about everywhere, which makes it ideal for live
remotes. But public Internet can also be erratic. You could be lucky enough to get a good connection,
but it might deteriorate during your broadcast. What to do? Cross your fingers and hope for the best? Or
reduce your bit rate, sacrificing audio quality in hopes of making it through your show?
With Z/IP ONE (the “Z/IP” stands for “Zephyr IP”), you don’t have to compromise audio quality for a solid
connection. Z/IP ONE helps you get the best possible quality from public IP networks and mobile data
services — even from connections behind NATs and firewalls. Telos® collaborated with Fraunhofer (the
developers of MP3 and many AAC breakthroughs) to develop a unique coding control algorithm that
adapts to changing Internet conditions on the fly, helping you maintain quality and stability.
We call it ACT, short for Agile Connection Technology, and only Telos has it. Using ACT to sense and
adapt to the condition of your IP link, Z/IP ONE delivers superb performance on real-world networks.
ACT adapts dynamically to minimize the effects of packet loss and jitter. When the bits are flowing
smoothly, you’ll benefit from the lowest possible delay and the highest possible fidelity. If congestion
starts to occur, Z/IP ONE automatically lowers bit rate and increases buffer length to keep audio flowing
at maximum quality. You’ll get reliable audio even when network conditions are unpredictable — and you
won’t need to fiddle with settings or codecs to do it.
To make certain your remote broadcast has excellent audio quality even when IP connections are notso-excellent, Telos engineers employed AAC-ELD (Advanced Audio Coding-Enhanced Low Delay) to
produce excellent fidelity at low bitrates, with nearly inaudible loss concealment and very little delay.
Standard high-performance codecs are a part of the Z/IP ONE toolkit as well, such as AAC-HE, AAC-LD,
MPEG4 AAC-LC, MPEG2 AAC-LC, G.711, G.722 and even linear PCM. And if apt-X is part of your codec
cache, you can add it to your Z/IP ONE as a small extra-cost option.
It’s from Telos, so of course you expect that Z/IP ONE will be easy to set up and easy to use. And it is
— the front panel controls are intuitive and friendly, and the built-in Web server makes short work of
configuration or remote control via any PC with a Web browser. And our exclusive worldwide Z/IP Server
service, free to Z/IP owners, lets you easily get around NATs and network firewalls for fast connections
to your favorite locations. For even more flexibility, Z/IP ONE can connect to third-party apps such as
LUCI LIVE and LUCI LIVE Lite to receive on-the-go reports from smartphones and tablets.
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TELOS | Z/IPONE
Around back, you’ll find analog and AES3 XLR ins and outs, a Livewire LAN port for quick connection to
Axia® networks, and a separate WAN port for safe connection to “the outside world.”
Z/IP ONE is also wireless-capable and connects natively to IP networks via Wi-Fi. A parallel port is
provided for end-to-end, time-aligned GPIO contact closures; Z/IP ONE can also transport RS-232 serial
data (using an inexpensive USB-to-Serial adaptor cable), synchronized with audio delivery — useful for
RDS/RBDS data, as well as other serial data, at up to 9600 bps.
SPECIFICATIONS
Conformance and Compatibility
• Conforms to N/ACIP (Open) Standards. Fully supports Session Initiation Protocol 2.0 (SIP). Compatible
with TCP, UDP, DNS, Zephyr Xstream®, Uncompressed PCM and other Internet Protocols.
Codecs
• SIP: G.711, G.722, MPEG Layer2, MPEG AAC, MPEG 4 AAC LC, MPEG 2 AAC LC, Linear PCM, MPEG
AAC-Enhanced Low Delay (ELD), High Efficiency AAC.
• Optional: apt-X Enhanced ® from CSR.
Connections
Analog
• 1x Stereo input, presented on two XLR-F connections
• 1x Stereo output, presented on two XLR-M connections
Livewire
• 1x 100BASE-T connections, presented on RJ-45
AES/EBU
• 1x Stereo Input, presented on one XLR-F connection
• 1x Stereo Output, presented on one XLR-M connection
Network
• 2x 100BASE-T connections, presented on RJ-45 (1x LAN, 1x WAN)
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TELOS | Z/IPONE
USB
• 2x A-Type, Female
Parallel (GPIO)
• 1x DB25, Male
Audio:
Analog Line Inputs:
• Input Impedance: 6K Ohm differential
• Input Range: Selectable, Line (+4 dBu nominal), Microphone (-50dBu nominal)
• Selectable Phantom power
Analog Line Outputs:
• Output Impedance: 50 Ohm differential
• Output Clipping: +22dBu
Digital Audio Inputs And Outputs
• Reference Level: +4 dBu (-20 dB FSD)
• Impedance: 110 Ohm, balanced
• Signal Format: AES3 (AES/EBU)
• AES3 Input Compliance: 24-bit with sample rate conversion
• AES3 Output Compliance: 24-bit
• Digital Reference: Internal (network timebase) or external reference 48 kHz, +/- 2 ppm
• Internal Sampling Rate: 48 kHz
• Input Sample Rate: 32 kHz to 192 kHz
• Output Sample Rate: 48, 44.1 or 32 kHz, or “sync to input” (auto-matches rate and clock from
AES/EBU input)
• A/D Conversions: 24-bit, Delta-Sigma, 256x oversampling
• D/A Conversions: 24-bit, Delta-Sigma, 256x oversampling
Frequency Response
• Any input to any output: +/- 1dB 25– 20 kHz
Headroom
• 18 dB
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TELOS | Z/IPONE
Dynamic Range
• 87dB Unweighted
• 90 dB “A” Weighted
Total Harmonic Distortion + Noise
• < 0.03% @ +12dBu, 1 kHz Sine
Crosstalk Isolation
• > 80 dB
Power Supply AC Input
• Auto-ranging supply, 90VAC to 240VAC, 50 Hz to 60 Hz, IEC receptacle, internal fuse
• Power consumption: 14.2 Watts
Operating Temperatures
• 0-40 degrees C (32-104 degrees F), stirred air
Dimensions
• 19” (48.3 cm) standard rack mounting front panel
• 1.75” (4.5 cm) height, 6.5” (16.51 cm) depth
• Shipping Weight: 8 lbs. (3.62 kg)
• Shipping Dimensions: 24” x 14” x 6” (61 cm x 35.6 cm x 15.25 cm)
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TELOS | ZEPHYR XSTREAM
Zephyr Xstream® ISDN Codec
The Best Way To Hear From There®
OVERVIEW
Zephyr Xstream is the world’s leading ISDN codec, compatible with the widest variety of third party
codecs. Coding choices include MPEG4-AAC and AAC-LD, Layer 2, Layer 3 & G.722 coding for full-duplex
stereo operation of up to 20 kHz audio on a single ISDN line; broadcast quality mono audio at 15 kHz
or 20 kHz is possible on a single ISDN “B” channel or other 56/64 kbps channel. All Xstream models
feature professional balanced analog inputs/outputs, as well as Livewire® I/O for quick connection to
Axia® networks; AES/EBU I/O is standard on rackmount model. An ISDN TA with integral NT1 provides
worldwide ISDN compatibility without software changes. Remote Control is possible over RS-232 or
Ethernet. Available in rack-mount version and portable version which incorporates a 4-channel stereo
DSP mixer with selectable digital processing by Omnia®.
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TELOS | ZEPHYR XSTREAM
FEATURES
• Ethernet ports for remote control via LAN or WAN, and for connection to your Livewire AoIP networks.
Bring audio from any codec anywhere in the world directly to your Axia network.
• Auto Receive mode quickly determines the correct coding algorithm for incoming audio streams.
• MPEG AAC (Advanced Audio Coding) permits true CD-quality stereo transmission with a connection
speed of just 128 kbps.
• Low-Delay MPEG AAC-LD coding delivers crystal-clear audio quality and greatly reduced encoding
delay for smooth, natural bi-directional remotes.
• MPEG Layer-3 coding for compatibility with the largest number of third-party codecs. When using
MPEG Layer-3, a unique Dual Receive mode allows reception of independent audio streams arriving
from two distant ISDN lines – great for bilingual broadcasts.
• Exclusive Error Concealment technology prevents occasional network glitches from being heard.
• RS-232 and 8-input, 8-output parallel ports provide ancillary data and bi-directional contact closures.
• Hand-in-glove operation with companion Zephyr Xport® portable codec for reception of 15kHz audio
using a POTS field connection.
• V.35/X21 option allows connection to serial synchronous data equipment, for use with dedicated lines,
Switched 56 circuits or satellite services.
• N/ACIP compliant for compatibility with the widest range of ISDN codecs.
• Convenient ISDN Voice Telephone Mode allows placement of standard G.711 phone calls from your
Zephyr Xstream.
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TELOS | ZEPHYR XSTREAM
IN DEPTH
Advanced caller management and superior sound
The Telos Zephyr is the best-loved broadcast codec in the world, and for good reason: Zephyr saves you
time and money. A Zephyr Xstream at your studio becomes a “universal codec,” connecting with every
popular ISDN codec for full-duplex, 20kHz stereo audio. And in the field, Zephyr Xstream is a powerful
remote tool, with intuitive step-by-step operation, context sensitive help, and a simple user interface
that eases operation for non-technical personnel. Zephyr pioneered the concept of the ISDN codec —
which is why you’ll find more Zephyrs in studios and remote kits around the world than any other codec.
Zephyr Xstream has a huge range of standard MPEG coding options, which include MPEG Layer-3 and
MPEG AAC for indistinguishable source-from-input audio at only 128 kbps. Zephyr Xstream can also
be used for LAN and WAN IP streaming of MP3 or AAC over properly managed networks. Zephyr’s AAC
coding includes error concealment to inaudibly recover from a lost packet or two, and an adjustable
packet jitter buffer allows you to easily accommodate different networks.
There are two Zephyr Xstream models tailored to fit your needs: The standard rack-mount Xstream,
and the portable Xstream MXP, a ruggedized portable version with built-in DSP mixer and Phantom
microphone power to help reduce field equipment inventory and setup times. All have standard Analog
and Livewire I/O, and a built-in terminal adapter with integral NT1 for worldwide compatibility without
software changes. The studio version also has standard AES/EBU I/O, and the portable Xstream features
a DSP-based AGC/limiter with Omnia audio processing and selectable presets for music & voice.
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TELOS | ZEPHYR XSTREAM
Ready for your rack
The Zephyr Xstream rackmount version is a full-featured ISDN transceiver that’s become the “gold
standard” for ISDN codecs around the world. In fact, Zephyr may be the most popular digital broadcast
product ever, with tens of thousands in service at radio and TV stations, recording and voice-over studios
everywhere. The front panel has a backlit display screen with a friendly, logical control structure and Fast
Access Menu Keys to quickly call up system information and settings. Other controls include meters for
send-and receive-audio levels, a dialing keypad, and a front-panel headphone jack with level control for
convenient direct monitoring. Zephyr Xstream also includes an Auto-Dial function with storage for up
to 100 stored Preset Numbers — each with its own bitrate and transmit/receive settings. 30 Location
settings permit quick recall of ISDN line and audio settings for your most commonly visited remote sites.
On the rear panel you’ll find standard balanced analog I/O with auxiliary unbalanced inputs, presented on
combination XLR/TRS connections. AES/EBU I/O is standard, with a separate AES sync input. Remote
control is supported using either the 100BASE-T Ethernet port or the RS-232 port. There are also eight
bi-directional inputs/outputs for end-to-end contact closure emulation. The built-in ISDN Terminal
Adapter is compatible with telcos worldwide. And Zephyr Xstream also has native Livewire support, for
one-cable connection to Axia AoIP networks.
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TELOS | ZEPHYR XSTREAM
Ready for the road
The portable Zephyr Xstream MXP has all the features found in the rackmount Zephyr Xstream, plus a
digital four-channel stereo mixer with two local mixes, inside a road-ready case designed for the rigors
of on-the-go broadcasting. The rugged shock-resistant case helps prevent bumps and bruises, and an
included flip-up metal stand lets you tilt the unit up for the best viewing angle on desktops or whatever
handy surface you’re broadcasting from. The alpha-numeric dial pad also generates DTMF tones for
navigation through voice menu systems.
Zephyr Xstream MXP’s four-input stereo DSP mixer directly feeds its codec section; mic/line switchable
inputs with pan also include a preset mic limiter & AGC processing by Omnia; inputs 1 & 2 have
switchable 48-volt Phantom power. There’s a front panel headphone jack for Local Mix 1 that
monitors either Send or Receive audio, or a mixture of the two. Local Mix 2 has separate front-panel
controls for the three rear-panel headphone jacks, plus a pair of balanced XLR line outputs to feed
guest phones or monitors.
Around back, the Xstream MXP differs from its rackmount brother by its inclusion of 4 input connections,
plus headphone/monitor outputs. Both Zephyr Xstream models are fan-free for silent operation.
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TELOS | ZEPHYR XSTREAM
SPECIFICATIONS
General
• Full duplex, high-fidelity codec using MPEG-2 AAC, MPEG-4 AAC-LD MPEG-2 Layer-3, MPEG-2
Layer-2, AACPlus, and G.722; fully compliant with international standards.
Frequency Response
• 20 - 20kHz @ 48kHz fs (+0/-1dB, swept sine tone procedure)
• AAC all modes except Stereo 64: 20-19,800Hz at 48kHz fs., 20-15,000Hz at 32kHz fs.
• AAC Stereo 64: 20-10,000Hz at 48kHz fs., 20-7,000Hz at 32kHz fs
• AACPlus mono (for use reception from the Xport): 20-15,000 Hz 48kHz fs
• Layer 3 all modes: 20-16,000Hz at 48kHz fs., 20-15,000Hz at 32kHz fs
• Layer 2 mono, dual mono: 20-7.8kHz/9.8kHz
• Layer 2 mono 20-8.6 kHz at 24 kHz fs.
• Layer 2 joint stereo: 20-20,000Hz at 48kHz fs. 20-15 kHz at 32kHz fs
• G.722: 20-7,500Hz.
THD+N
• Audio loopback, 48kHz fs, analog I/O, input at 1kHz +20dBu: 0.004%
Dynamic Range
• A Weighting, AAC, Layer-3 or 2 end-to-end: 101dB typical
Send Input
• Active balanced with RF protection.
Zephyr Xstream:
• LINE: Settable to -11 or +4dBu (or -15 to 0 dBu) nominal level
• Clip point: 18 dB above chosen nominal level.
• Impedance: > 10K Ohms (x2)
• Connector: XLR female/quarter-inch TRS combo connector.
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TELOS | ZEPHYR XSTREAM
Zephyr Xstream MX and MXP:
• LINE: -11 or +4dBu nominal level (switchable).
• MIC: Accepts -65 to -24 dBu in 2 ranges (switchable). Mic impedance </= 1000 Ohms
• Clip point: 15 dB above chosen nominal level.
• Impedance: Line > 10K Ohms (x2)
• Connector: XLR female/quarter-inch TRS combo connector.
Limiter
Zephyr Xstream MXP:
• Internal DSP-based AGC/limiter with Omnia® audio processing. Includes presets for music & voice,
selectable per channel.
Zephy Xstream:
• Analog soft-clipper prevents A/D converter overload without loss of dynamic range.
Line Bit Rates (ISDN)
• 56 or 64kbps per channel, front panel selectable.
Bit Rates (V.35/X.21)
• 56, 64, 112 (imuxed), 128 (imuxed), 96, 128, 256, 384 kbps front panel selectable.
Receive Output
• Active differential.
• Level: Front panel selectable for -10 or +4dBu, nominal.
• Impedance: < 33 Ohms (x2)
• XLR male
AES/EBU Digital I/O (rackmount version only)
• Sample rates supported: 32, 44.1 and 48kHz
• Rate conversion: Input and output independently selectable. Can be bypassed.
• Input clock: From external source or Telco clock.
• Output clock: From transmission sample rate, external source, or AES/EBU input.
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TELOS | ZEPHYR XSTREAM
Inverse Multiplex/Demultiplex
Internal channel splitting/combining of two network channels for stereo modes.
• AAC: Telos Zephyr protocol.
• AAC-LD: Telos Zephyr protocol.
• Layer-3: FHG/Telos Zephyr (Buchta) protocol.
• Layer-2: CCS CDQ™ protocol compatible.
Optional V.35/X.21 Direct Digital Interface
• Two ports, both V.35/X.21. Automatically selected when the appropriate cable is connected.
ISDN Interface
• Compatible with National ISDN, AT&T 5ESS custom, Northern Telecom DMS-100 custom, Siemens
EWSD, INS 64(Japan) and EURO-ISDN (ETS-300). Compatibility and approval pending in some
countries; contact Telos for current status.
LAN Interface
• 100BASE-T Ethernet port using RJ-45 connector. Full Duplex Supports TCP/IP (HTML, Telnet and FTP).
ISDN Voice Telephone Mode
• Two channels using G.711 standard, μ-Law or A-Law. 300–3,400Hz. DTMF signaling provided
(CCITT standard).
• Remote Control and Ancillary Data
• RS-232 9-pin D-Sub female (DCE): Asynchronous; 8 data, no parity, 1/2 stop, 2400-57,600 bps.
• 100BASE-T Ethernet port using RJ-45-style connector using Telnet or web browser (HTML).
Regulatory
North America: FCC and CE tested and compliant, power supply is UL approved.
Europe: Complies with the European Union Directive 2002/95/EC on the restriction of the use of certain
hazardous substances in electrical and electronic equipment (RoHS), as amended by Commission
Decisions 2005/618/EC, 2005/717/ EC, 2005/747/EC (RoHS Directive), and WEEE.
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TELOS | ZEPHYR IPORT PLUS
Zephyr® iPort PLUS
Multi-CODEC Gateway
16 Stereo Codecs in a Livewire® Gateway
OVERVIEW
Zephyr iPort PLUS is a networked multi-codec gateway that enables transport of multiple channels of
stereo audio across any QoS-enabled IP network, including T1 and T3 connections and private WANs
with MPLS – perfect for large-scale distribution of audio to single or multiple locations.
Zephyr iPort PLUS is the workhorse of codecs, configurable as eight stereo bi-directional MPEG codecs,
or for encode / decode of up to 16 uni-directional stereo streams. Zephyr iPort PLUS connects to Axia®
IP-Audio networks using a single CAT-6 cable for all I/O. Don’t have a Livewire network yet? Pair Zephyr
iPort PLUS with Telos Alliance® xNode audio interfaces for use as a standalone multiple-stream codec.
Coding algorithms include AAC, AAC-LD, HE-AAC (plus v2), MP2, MP3, linear, and optional aptX®
Enhanced*. Bit rates range from 24 to 320 kbps for MPEG codecs, plus standard fixed rates for aptX
and linear to over 2 Mbps. In addition, iPort offers dual, parallel-path end-to-end streaming for ultrareliability and redundancy. For network operators, a unique Content Delay feature allows independent
local storage and scheduled delayed playout of any or all coded audio channels for up to six hours.
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TELOS | ZEPHYR IPORT PLUS
FEATURES
• Distributes multiple channels of coded audio between broadcast facilities over QoS-enabled IP links.
• Configurable as a CODEC with 8 bi-directional channels, each with GPIO and PAD — or, as a
16-channel stereo encoder or 16-channel stereo decoder.
• 8 PCM Stereo channels are available for use simultaneously alongside CODEC channels, (dependent
upon available bandwidth).
• Can also deliver streaming audio channels for Internet transmission via SHOUTcast, Steamcast or
compatible stream replication server.
• Wide choice of genuine Fraunhofer codecs, including Standard AAC, high-efficiency AAC-HE (aacPlus),
AAC-HEv2, low-delay AAC-LD, and MP3, with a choice of bit rates from 24 kbps to 320 kbps, definable
per stream.
• Optional aptX Enhanced audio coding may be ordered at time of purchase or added later, as desired.
• When used as part of a Livewire network, allows audio from remote facilities to be used as if they
were local sources, with associated logic and control.
• Two 5-input Virtual Mixer (VMIX) channels each allow combining and mixing of up to 5 networked
Livewire audio streams on a single channel.
• Eight Virtual Mode (VMODE) channels allow audio to be split into left/right channels, summed L+R,
and more, prior to encoding and transmission.
• Content Delay option enables delayed playout of any or all selected receive audio channels, along
with time-synchronized ancillary data, for up to six hours. Each playback delay time is independently
configurable on a per-channel basis, making Zephyr iPort PLUS ideal for network operators, program
distribution networks, or delayed playout of received audio at network-affiliated stations.
• Remote control/configuration via any computer with a standard Web browser.
• Separate LAN and WAN ports help ensure network security.
• Fanless, convection-cooled DSP-powered platform with dual-redundant, auto-switching
powersupplies for maximum uptime. Power supply modules are field-replaceable in minutes.
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TELOS | ZEPHYR IPORT PLUS
IN DEPTH
Powerful, advanced program distribution and facility connection.
If your facility is like most, rack space is a precious commodity. That’s why Telos® engineers invented
Zephyr iPort PLUS, a sophisticated multiple-CODEC device that saves you money and rack space by
housing 16 broadcast-quality stereo codecs in one 2RU device.
A pair of Zephyr iPort PLUS on each end of a QoS-controlled IP link can send and receive 8 channels
of bi-directional stereo MPEG audio. Or, use iPort as a one-way “push” link to encode and deliver 16
channels of broadcast-quality one-way audio to a remote destination. With its ability to send multiple
MPEG channels over IP connections, Zephyr iPort PLUS is perfect for audio transmission over VPNs,
satellite links, Ethernet radio systems, and Telco or ISP-provided QoS-controlled IP services such as T1,
T3 or OC-3 links.
You can use iPort for studio-to-transmitter links, network distribution systems, multi-channel links to
remote studios. Install a QoS-enabled IP link between two studios with Axia Livewire networks, put an
iPort at each end, and you can pass audio and GPIO between locations as if they were just next door.
Paired with an appropriate server, you can even use Zephyr iPort PLUS to generate multiple channels of
MP3 or AAC coded audio for Internet streaming, broadcasting to mobile phones, and audio distribution
systems.
Finally, Zephyr iPort PLUS’ exclusive Content Delay option (available at extra cost) adds hardware and
software that enables delayed playout of select received audio channels. Associated GPO and ancillary
data is likewise delayed and synchronized with audio. Delay any or all coded audio channels up to six
hours; each channel’s delay time is independently configurable.
The Zephyr iPort PLUS rear panel is remarkably simple, thanks to the use of Livewire AoIP I/O. A single
Ethernet cable is all that’s needed for all inputs, outputs, GPIO and remote control. Uncompressed
24-bit/48kHz audio goes in from your network via Ethernet; compressed MPEG streams go out on the
same cable — eliminating expensive, space-consuming converters and connectors. Or, use the separate
WAN connection to send your audio over an outside network.
If you don’t have an Axia network yet, that’s no problem — just pair Zephyr iPort PLUS with Telos VX
analog or digital audio interfaces, or Telos Alliance xNodes, to make a standalone high-density audio
codec package.
Zephyr iPort PLUS streams sound fantastic, thanks to our long-standing relationship with Fraunhofer IIS,
the inventor of MP3 and co-inventor of AAC. The encoding algorithms inside iPort are genuine FhG, not
no-name knockoffs. A full range of state-of-the-art codec types and bitrates are supported; the highestquality implementations possible, run by a powerful Intel floating-point processor. Choose AAC-LD for
delay-sensitive applications, AAC-HE and AAC-HEv2 for low bitrate requirements, standard MPEG AAC
for best quality and resilience to packet loss at higher bitrates, MP3 and MP2 for legacy applications.
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You’d expect all this to cost a lot, but it doesn’t: we built Zephyr iPort PLUS on a single industrial
motherboard, rather than the usual “multiple DSP cards in a frame” approach. Together with the
Livewire-only audio interface, Zephyr iPort PLUS delivers more power than a legacy cardframe design, at
only a fraction of the cost.
SPECIFICATIONS
Audio
Zephyr iPort PLUS has no native audio I/O, operating on streams provided by attached Livewire audio
devices. All audio specifications below are representative of Axia Livewire audio interfaces.
Analog Line Inputs
• Input Impedance: >40 k ohms, balanced
• Nominal Input Range: Selectable, +4 dBor -10dBv
• Input Headroom: 20 dB above nominal input
Analog Line Outputs
• 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 dB(-20 dB FSD)
• Impedance: 110 Ohm, balanced (XLR)
• Signal Format: AES3 (AES/EBU)
• AES3 Input Compliance: 24-bit with selectable sample rate conversion, 32 kHz to 96 kHz input
sample rate capable.
• AES3 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
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Frequency Response
• Any input to any output: +/- 0.5 dB, 20 Hz to 20 kHz
Network
• 1 LAN port, 1 WAN port; 100/1000BASE-T Ethernet interfaces.
Codecs
• Standard AAC, high-efficiency AAC-HE (aacPlus), AAC-HEv2, low-delay AAC-LD, MP3, MP2. Optional:
apt-X® from CSR.
Power
• Dual-redundant internal auto-ranging power supplies, 90 – 132 / 187 – 264 VAC, 50Hz/60Hz.
• Power consumption: 100 Watts.
Regulatory
North America: FCC and CE tested and compliant, power supply is UL approved.
Europe: Complies with the European Union Directive 2002/95/EC on the restriction of the use of certain
hazardous substances in electrical and electronic equipment (RoHS), as amended by Commission
Decisions 2005/618/EC, 2005/717/ EC, 2005/747/EC (RoHS Directive), and WEEE.
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TELOS | VX PRIME
Telos VX Prime
Big Performance for Small Facilities
OVERVIEW
Telos VX talk-show systems are the world’s only true VoIP-based broadcast phone systems. The
VX Prime gives you incredible operational power, flexible, adaptable workflows, and superior audio
quality—a powerful broadcast phone solution that’s economical enough for stations with just two or
three studios. VX Prime connects to VoIP-based PBX systems and SIP carriers to take advantage of lowcost and high reliability service offerings. VX Prime makes it easier than ever for talent to have complete
mastery of their callers. No other broadcast phone system delivers the power of VoIP to the broadcast
studio like Telos VX. With VX Prime, the world’s leading broadcast phone system is now available to
those with smaller budgets, giving you big studio sound at a small studio prices. Simply put, you’re
paying for the capability you need, and nothing extra.
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Why VX Prime?
Cost-Efficient Way to Upgrade to IP
• Lower cost alternative to VX Broadcast System
• Potentially Save thousands monthly on expensive ISDN/POTS lines
• Ideal for Smaller Studios (2-4 Stations) & Smaller Budgets
Audio Quality
• Native support of G.722 “HD Voice” codec
• Fifth-generation Telos Adaptive Digital Hybrid on every line for cleanest, clearest caller audio
• Smart AGC ensures consistent caller audio levels
• Digital Dynamic EQ (DDEQ) by Omnia adjust EQ automatically to ensure call-to-call consistency
and the best intelligibility
Simple Setup
• Connects to your existing Livewire network with a single Ethernet cable
• Non-Livewire studios can use Telos Alliance Mixed Signal xNode for audio and GPIO connectivity
to studio consoles
• Provides phone hybrids for each of your studios without need for any additional wiring or
physical audio connections
Flexible Use
• No restriction to the number of SIP lines or phone numbers that can come into the system
• Eight Fixed Hybrid/Faders (not expandable)
Support
• Industry-leading 24/7 support
• Telos standard 5-Year Warranty
• Free Xscreen Call Screening Software from Broadcast Bionics
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FEATURES
Features At A Glance
• A true VoIP telephone system designed and built specifically for broadcasting; VX Prime is ideal for
small to medium studios with two to four stations.
• SIP call-handling throughout—no internal conversion to analog call handling like some other so-called
“VoIP” systems.
• Standards-based SIP interface integrates with Asterisk open-source SIP phone servers and most
VoIP-based PBX systems to allow transfers and common telco services for business and studio
phones.
• Standard Ethernet backbone provides a common transport path for both studio audio and telecom
needs, resulting in cost savings and a simplified studio infrastructure.
• System capacity of eight hybrids. Each call placed on the air receives a dedicated hybrid for unmatched
clarity and superior conferencing.
• Native Livewire integration—one connection integrates caller audio, program-on-hold, mix-minus,
and logic directly into Axia AoIP consoles and networks.
• Connect VX systems to any third-party radio console or other broadcast equipment using available
Telos Alliance Mixed Signal, AES/EBU, and GPIO xNodes. xNodes feature 48 kHz sampling rate and
studio-grade 24-bit A/D converters with 256x oversampling.
• Powerful dynamic line management enables instant reallocation of call-in lines to studios requiring
increased capacity.
• VSet phone controllers with full-color LCD displays and Telos Status Symbols present producers and
talent with a rich graphical information display. Each VSet features its own address book and call log.
• Drop-in Call Controller modules can integrate VX phone control directly into your mixing consoles.
• Included XScreen Lite screening software with built-in soft-phone allows a “phone” connection on any
networked PC. Integrated recorder/editor simplifies recording of off-air conversations.
• Clear, clean caller audio from fifth-generation Telos Adaptive Hybrid technology, including Digital Dynamic
EQ, AGC, adjustable caller ducking, and send- and receive-audio dynamics processing by Omnia.
• Support for G.722 “HD Voice” codec enables high-fidelity (7 khz) phone calls from SIP telephone sets
and softphones.
• Works with POTS, T1/E1, ISDN and SIP Trunking telco services for maximum flexibility and cost
savings, via standard Telco gateways.*
* Due to the wide variation in how traditional phone service can be delivered, and the complexities that
can be involved in converting those services to SIP, we really want to talk with you about your system
design before you order. Telos has VX System engineers standing by to help you draw up a configuration
that will ensure your VX purchase will perform to your expectations when using traditional POTS and
ISDN lines.
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IN DEPTH
This Is Where It All Started
Steve Church founded Telos Systems in 1985. As both a talk-show host and radio group Technical
Director, Steve was only too familiar with the frustrations of “bad phones” and even less responsive
equipment manufacturers, so he set about eliminating the technical problems that plagued radio call-in
segments. In 1984, he invented the Telos 10, the first DSP-based telephone-to-broadcast interface
system—allowing radio stations to significantly improve the technical quality of call-in segments. The
overwhelming response to Steve’s economical and technically elegant solution to a nagging problem
provided the spark from which Telos was born.
A lot’s happened since then. Telos pioneered the use of MPEG Layer 3 coding in the revolutionary Zephyr
ISDN codec. We produced the first hardware MP3 streaming encoder for broadcast. We developed the
world’s first “whole-plant” broadcast phone system. We invented the IP-networked radio console, and
then integrated broadcast phones into that network via Ethernet.
Telos has grown steadily since our initial production run of 25 Telos 10 units in 1985! With tens of
thousands of systems in the field, it’s now is hard to find a broadcast facility in the world without at least
one piece of our gear. Our organization, now called The Telos Alliance, includes the Omnia Audio, Axia
Audio 25/Seven Systems, Minnetonka Audio and Linear Acoustic brands, and our R&D department—
the largest research team in broadcasting— continues to develop innovative audio products for radio
and television broadcasting, telephony, and the Internet.
VX Prime Broadcast VoIP Phone System
Telos VX marries the flexibility and capabilities of IP networks to the remarkable power of today’s digital
signal processing, and brings the benefits to broadcast facilities. With a VX system, you can move and
share lines between studios at the touch of a button. Choose VX Prime, an eight-hybrid system that
brings VoIP flexibility to medium and small facilities without breaking the bank.
VX systems are naturally flexible, naturally powerful. Your broadcasts benefit from superb, crystal-clear
caller audio while callers hear clean, intelligible audio from your console. VX systems are surprisingly
cost-effective, even when deployed in single-station facilities.
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Why VoIP For Broadcast?
VoIP has taken the business world by storm, increasing the flexibility of office phone systems and PBXs
while simultaneously lowering maintenance and equipment costs. In fact, most Fortune 500 companies
have replaced their old PBX systems with VoIP for just these reasons.
VoIP is a natural for broadcasters, interconnecting the phone system with audio interfaces, phone sets,
console controllers, and PCs running screening software by way of efficient, low-cost Ethernet. Using
VoIP, you can finally share phone lines among multiple studios and route caller audio anywhere in your
facility, easily, and instantly. Got a hot talk-show that suddenly needs more lines in a certain studio? Just
a few keystrokes at a computer and you’re ready no delays, and no cables to pull. VX systems can even
interconnect with your business office’s VoIP PBX to allow easy call transfers.
Reduced Cost. Increased Flexibility.
The use of sophisticated, modern IP networking for Telos VX Prime allows rich communication between
devices. For example, caller information entered by a producer is displayed on the studio phone’s color
LCD. Caller audio is available on studio PCs for easy recording. Operators at mixing consoles can directly
control line switching without diverting their attention from the board. The result? Talk shows that run
like clockwork, sound better, and flow without errors.
This standards-based VoIP architecture helps you save money, too, by widening your choices in telco
providers. Most carriers now offer VoIP services using the SIP protocol, which can deliver substantial
savings to stations that need any number of lines. (You can also connect to traditional T-1/PRI, POTS or
ISDN phone lines using open-source Asterisk-based phone servers.)
But VX systems don’t stop at providing the benefits of VoIP—they also carry the broadcast-phone
technology expertise of Telos.
Every incoming line has its own fifth-generation Telos Adaptive Digital Hybrid, our most advanced
ever—packed full of technology engineered to extract the cleanest, clearest caller audio from just about
any phone line, even cellular calls. Multiple lines can be conferenced with superior clarity and fidelity.
Smart AGC ensures consistent caller audio levels. And calls from mobile callers using SIP clients on
their smartphones benefit from native support for the G.722 “HD Voice” codec, improving caller speech
quality and intelligibility.
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VX System Components
VX Prime Engine
The heart of any VX system is the Engine. The fixed-capacity VX Prime system is powered by a 2RU
rack-mount Engine with enormous processing power: the VX Prime Engine provides all the call control
and audio processing needed for your entire on-air phone system.
With VX Prime, you are equipped with eight high-performance VoIP hybrids, to support multiple lines of
concurrent on-air phones for from two to four studios (dependent upon configuration).
Each VX Prime Engine features two Gigabit Ethernet ports, a high-density, cost-effective interface to both
telephone lines and studio audio via proven Livewire Audio over IP (AoIP). VX systems are web-based, so
remote control and configuration are easy—engineers can work from anyplace they can get online.
Call workflow for VX users is sophisticated and flexible. No matter which system you choose, 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 VSet phone controllers and console drop-in modules. Each studio may
be configured with its own Program-on-Hold as well.
The processing power of the VX Engine provides sophisticated DSP hybrids for every line, allowing
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. Call ducking and host override are part of the VX audio
toolkit as well.
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You’ll notice that there are no audio I/O or telco ports on VX Engines themselves. That’s because they’re
meant for fast connection to Livewire AoIP systems; using Livewire, all I/O is handled via Ethernet. The
Livewire network supports a wide variety of peripherals such as Axia audio consoles, VSet phones,
PC-based screener applications, console-integrated controllers, and more. SIP servers and telecom
providers connect through a dedicated WAN Ethernet jack for routing simplicity and easy maintenance.
For traditional phone services, VX works seamlessly with open-source Asterisk SIP servers, and
most SIP PBX’s. Telos VX experts speak fluent Asterisk, and are ready to assist you in specifying
and configuring an installation to suit your studio’s requirements. VX also works with standard telco
gateways to connect to T1/E1, ISDN, and POTS providers. And, if you already have a VoIP-based PBX or
SIP endpoint service, VX systems can work with those as well.
VSet12
The Telos VSet12 phone is beautifully designed, with a friendly LCD color display that uses exclusive
Status Symbols to let talent know what's going on instantly. VSet12 works with up to 12 phone lines;
the info-rich display provides caller ID for each line, along with time ringing-in or on-hold, and even
screener comments from the screening software applications.
VSet12 gives talent unprecedented flexibility. You can map groups of lines to a single fader, making it
simple to take a queue of calls to air sequentially. One-touch controls let talent step through queued
calls, “busy out” incoming lines, lock calls on-air to prevent unintentional disconnection of a VIP. Telosexclusive “Next Call” key speeds workflow for producers, screeners, and talent. But because VX systems
provide a hybrid per line, much more functionality is unlocked: You can now spread multiple calls over a
number of faders, using one for each call so that operators can control each line’s level individually. You
can hard-assign individual lines to fixed faders, such as for VIP calls. A built-in address book and call
history log round out VSet12’s features.
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VSet6
VSet6 is a six-line phone controller for VX systems. Like the VSet12, it has a bright, attractive LCD color
display with Status Symbols that feed talent instant information about line and caller status and controls
that enable talent to step through queued calls, busy incoming lines, lock calls on-air, activate the dump
button on a profanity delay, and more.
VSet Phone Controls
The LCD displays deliver detailed line status, caller information, caller ID, time ringing-in or on-hold, and
even comments entered in screening software applications. Shown above are a few of the attractive,
instantly-understandable Status Symbols that help talent run tight, mistake-free shows.
Each VSet phone has its own web server for easy remote configuration and software upgrades, and
flexible power options include PoE (Power over Ethernet) from a Telos-approved switch or Axia xSwitch,
or the included in-line power injector.
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On-Console Control
Whether calls are live or pre-recorded, interviews or audience participation, one thing’s certain: phone
segments are an integral part of today’s fast-paced radio. But up to now, the phone system was
separate from the on-air console; audio was shared, but little else. Wouldn’t it be great if talent could
take control of phones without ever having to divert their attention from the board? They can: the Axia
Console Controller provides the ideal way to integrate broadcast phones into the on-air console—the
control center of every studio.
There are plenty of advantages to melding phones with consoles. Like ease of installation: IP-Audio
consoles with built-in phone controllers don’t need any additional wires or connections. Their control
signaling, caller audio, and backfeeds ride on the network connection that’s already there. Bringing caller
audio into the IP-Audio domain makes it routable like any other audio source. You can even dynamically
conference multiple lines using just a single fader.
VX systems connect directly to Axia Fusion, Element, iQ and Radius mixing consoles using Livewire+™
AES67 IP-Audio to eliminate the cost and complexity of old-style inputs, outputs, and mix-minuses.
Multiple phone lines— each with a dedicated hybrid—can automatically map to individual console
faders for complete control of caller audio. And users enjoy seamless console integration, with phone
controls right on the board so that talent can dial, answer, screen, and drop calls without ever diverting
their attention from the console. Information about line and caller status can be displayed right on the
console as well.
Drop-in VX control modules are available for use with other console brands, too.
Axia Call Controller
For VX clients with Axia Fusion or Element AoIP mixing consoles, the Axia Call
Controller module puts control of VX telephone systems right into the console.
The two-fader telephone control module features an integrated Telos Call
Controller with renowned Status Symbols™ visual call management, Transfer,
Drop, and Block All keys plus the Telos-exclusive “Next Call” key that allows
fast airing of pre-screened calls. The rotary Options Control knobs can be
programmed to trim source or fader gain when turned, and alphanumeric
channel displays give complete information on source assignment, channel
options, and more.
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VSet Call Controller
Want a VX system, but don’t have an Axia mixing console? No problem — Telos
provides VSet Console Controller electronics packages, which may be fitted to your
console using panels supplied by your OEM console provider or preferred third-party
fabricator. Like the VSet12 phoneset, the VSet Console Controller provides visual
line-status indicators and fast-take keys for selection and control of up to 12 callers,
along with standard controls such as Take, Drop, Hold and Busy keys, and the Telosexclusive “Next Call” key to speed workflow for producers, screeners, and talent. There’s
also a built-in keypad for on-console dialing of outgoing numbers.
Broadcast Bionics XScreen Call-Screening Software Included
XScreen software comes with every
VX Engine purchase and provides call
control, call screening, data capture,
and chat functionality enabling you to
quickly answer, screen, and route calls
using multiple PC clients. The cloudbased database keeps a log of calls
and provides further alert and directory
functionality.
XScreen can record and manage caller audio (Livewire systems only) and can additionally act as a
softphone for talking to and screening callers directly through a USB headset or soundcard on your
XScreen client PC. XScreen is available in free (Lite) and full (subscription) versions. When you install XScreen for the first
time you will receive a 90-day free trial license for the full version. After 90 days you can continue to use
the full version with an annual subscription, or use the reduced, Lite functionality free of charge. Please
download your XScreen software from www.xscreen2.com.
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VX Prime Interfaces
VX Audio and Logic Interfaces let you connect VX to any non-networked radio console or other broadcast
equipment, using multipurpose or AES/EBU interfaces. A GPIO Logic interface provides control logic
where it’s needed. Our new, space-efficient Telos Multipurpose Node offers a mix of mix of analog, AES,
and GPIO connections’ a perfect “utility node” for any VX system.
VX AES/EBU Audio Interface
The VX AES/EBU audio interface provides eight digital AES3 inputs and outputs, each on a separate RJ45 connector. Studio-grade performance specs, like 138dB of dynamic range and <0.0003% THD.
VX GPIO Logic Interface
Each VX GPIO logic interface has eight assignable logic ports. Each port contains five opto-isolated
inputs and five opto-isolated outputs, which can be associated with audio input peripherals and/or
output destination devices to provide machine start/stop pulses, lamp drives, and transport controls.
Once a port is configured to be associated with a particular device, it automatically activates with that
device.
Telos Alliance Mixed Signal xNode
Compact, AES67-compliant 9.5” x 11” third-generation AoIP interface with a mix of analog, AES and
GPIO connections. There’s one mic/line analog input (switchable), two analog line inputs (dedicated),
three analog line outputs, one digital AES3 input, and one AES3 output and two GPIO ports, each with
five opto-isolated inputs and outputs. Dual Ethernet ports allow connection to fully redundant networks.
It can even be powered by AC mains or PoE from Ethernet switches compatible with the IEEE 802.1af PoE
standard. 1RU height and half-rack width allows side-by-side mounting of two units in one rack space.
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The Power of IP Realized
With VX Prime, there’s no need for the maze of discrete cables once required by multi-line talk show
systems. All VX components are linked with standard Ethernet, so a single CAT-5 cable provides:
• Connection to the telco interface
• Line switching commands
• Data communication between the VX Engine and VSet12 phones
• Transport of caller audio to mixing consoles
• Return of mix-minus and program-on-hold audio to the caller
• Data messages (such as call notes and IM) between producer and talent
• Livewire audio for the recording of calls
• Transfer of recorded call files from the producer to the studio
Now… how many discrete cables does that save you from having to wire up?
Hook It Up Your Way: Non-Axia Installation Diagram
WAN PORT
LAN PORT
VX PRIME
ETHERNET SWITCH
MIXED SIGNAL XNODE
ASTERISK PBX, PBX, SIP PROVIDER, OR GATEWAY
DELAY UNIT
VSET12
VSET12
CONSOLE
CONSOLE
ROUTER
ORORROUTER
“LINE RINGING” LIGHT
XSCREEN
Got an Axia Livewire AoIP studio network? Great! Your new VX Prime phone system will plug right into it.
It’s the seamless integration of studio phones, mixing consoles, and routing network you’ve dreamt about.
Don’t have IP-Audio networking yet? Not to worry… VX will work with all console brands, networked or
not, using Telos Alliance xNodes, and the VX Call Controller drop-in controller for your console.
Telos VX systems are “facility wide” broadcast phone products. That means multiple studios, multiple
stations, multiple shows — with minimal hardware requirements. Telco is delivered via IP from your SIP
PBX, or through a dedicated IP circuit using SIP trunking. POTS, ISDN, or T1 phone service can be brought
in using an open-source Asterisk server or a standalone gateway device. Once connected, all line and
audio connectivity flows via Ethernet. The diagram above shows a typical studio with an analog mixer,
using a Telos Alliance Mixed Signal xNode to connect to the console and other broadcast equipment.
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Hook It Up Your Way: Axia Installation Diagram
LAN PORT
WAN PORT
POWERSTATION
VX PRIME
ASTERISK PBX, PBX, SIP PROVIDER, OR GATEWAY
LIVEWIRE-CAPABLE DELAY
“LINE RINGING” LIGHT
VSET12
VSET12
AXIA CONSOLE
XSCREEN
Installing a VX phone system in facilities already powered by Axia Livewire+™ AES67 networks requires
even less time and hardware than described above. The audio inputs and outputs used and produced
by VX are Livewire real-time audio channels and travel over the Axia AoIP network just like the rest of
your audio. Axia console GPIO ports can be used for “phone ringing” tallies or remote control of profanity
delay units.
VX Gives You Options
Broadcast Bionics
Broadcast Bionics offers PhoneBOX VX, a tailored-for-VX version of their original PhoneBOX software.
PhoneBOX VX gives VX users an amazing amount of information and a high level of control over the VX
system. There’s prize management, call editing, and recording, sophisticated visual talkback, including a
drag-and-drop database your show’s calls, plus a rich phonebook and visual warnings, tied to Caller ID,
for persistent or nuisance callers.
Find out more from www.phoneboxvx.com .
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NeoSoft
NeoSoft offers NeoScreener, a call management solution that interfaces Telos NX12, NX6, IQ6, VX, HX6,
2x12 and 2101 systems, allowing for line control and database lookup using caller ID. The solution can
interface to NeoWinners which is NeoGroupe's contest management software. It is designed for radio
and television stations that need to manage their flow of incoming phone calls.
NeoScreener also handles external inputs, like SMS, Website, iPhone. Database driven, it enhances the
phone-call workflow. With NeoScreener, call screeners can easily welcome calls and present them to the
Talent on a specific display. Visit www.neogroupe.com to learn more.
Arctic Palm CS Call Management
The CS Call Management package provides producers and talent with the tools to capture and control
callers while staying in touch with each other in a single Caller Control window. Designed for the VX VOIP
systems, both local and remote users are in constant communication. For more information, visit www.arcticpalm.com/CSScreener.htm.
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TELOS | VX PRIME
SPECIFICATIONS
System
• Maximum number of simultaneous calls on-air, VX Prime: 8 (more with conferencing)
• Maximum number of SIP numbers, VX Prime: 96
Audio Performance (Node)
Analog Line Inputs
• Input Impedance: >40 k ohms, balanced
• Nominal Level Range: Selectable, +4 dBu or -10dBv
• Input Headroom: 20 dB above nominal input
Analog Line Outputs
• 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) h 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
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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 Prime Engine
IP/Ethernet Connections
• One 1 Gigabit Ethernet via RJ-45 LAN connection (livewire)
• One 1 Gigabit Ethernet via RJ-45 WAN Connection (SIP provider)
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 (3 band)
• Ducker
• Sample rate converter
Power Supply AC Input
• Hot-swap capable dual-redundant internal auto-ranging power supplies. 90 – 132 / 187 – 264 VAC,
50Hz/60Hz. IEC receptacle, internal fuse.
• Power consumption: 100 Watts
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TELOS | VX PRIME
Operating Temperatures
• -10 degree C to +40 degree C, <90% humidity, no condensation
• Fanless, convection-cooled
Dimensions and Weight
• Rackmount, 2RU
• 3.5 inches x 17 inches x 15 inches
• 10 pounds
Studio Audio Connections
• Via Livewire Ethernet. Each selectable group and fixed line has a send and receive input/output.
• Each studio may be configured with its own Program-on-Hold input.
• Livewire-equipped studios take audio directly from the network.
• Telos Alliance xNodes are available for professional-level analog and AES3 connection breakouts for
clients without Livewire AoIP networking.
Telco Connections
• Audio: standard RTP. Codecs: G.711u-Law and A-Law, and G.722.
• Control: standard SIP endpoint
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TELOS | VX BROADCAST VOIP
VX Broadcast VoIP
The Whole-Plant Broadcast Talkshow System
OVERVIEW
VX is the world’s first VoIP (Voice over IP) talkshow system — a broadcast phone system that’s so
powerful, it can run all of the on-air phones for your entire plant, but economical enough for stations
with just two or three studios. VX connects to traditional POTS and ISDN telephone lines via standard
Telco gateways. But it also connects to VoIP-based PBX systems and SIP Trunking services to take
advantage of low-cost Internet-delivered phone services.
VX weds modern networking to the remarkable power of digital signal processing. VX uses Ethernet as
its connection backbone, significantly cutting the cost of phone system installation, maintenance, and
cabling. Ethernet is a powerful, yet simple way to share phone lines among studios and connect system
components. This also makes VX naturally scalable, capable of serving even the largest of facilities —
while remaining surprisingly cost-effective for even single stations with more modest needs.
Don’t have an IP-Audio network yet? No problem; optional Telos Alliance xNodes, like the Telos Alliance
Mixed Signal Node, break out audio into analog and digital formats, along with GPIO logic commands.
And with informative VSet phones, talent finds it easier than ever to take control of their callers, moving
and sharing lines between studios at the touch of a button.
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TELOS | VX BROADCAST VOIP
FEATURES
• World’s first VoIP telephone system designed and built specifically for broadcasting.
• Works with POTS, T1/E1, ISDN and SIP Trunking telco services for maximum flexibility and cost
savings.
• Standards-based SIP/IP interface integrates with most VoIP-based PBX systems to allow transfers,
line-sharing and common telco services for business and studio phones.
• Standard Ethernet backbone provides a common transport path for both studio audio and telecom
needs, resulting in cost savings and a simplified studio infrastructure. Connection of up to 100 control
devices (software or hardware) is possible.
• Modular, scalable system can be easily expanded to manage a network of up to 20 studios, each with
a dedicated Program-On-Hold input – truly a “whole-plant” solution for on-air phones.
• System capacity of up to 48 standard phone lines; supports up to 250 SIP numbers.
• Up to 16 hybrids. As many as 48 active calls (up to 4 per hybrid) can be placed on-air concurrently.
• Each call receives a dedicated hybrid for unmatched clarity and superior conferencing.
• Native Livewire® integration: One connection integrates caller audio, program-on-hold, mix-minus,
and logic directly into Axia AoIP consoles and networks.
• Connect VX to any radio console or other broadcast equipment using available Telos Alliance AES/EBU,
Mixed Signal, and GPIO xNodes. Audio interfaces feature 48 kHz sampling rate and studio-grade 24bit A/D converters with 256x oversampling.
• Powerful dynamic line management enables instant re-allocation of call-in lines to studios requiring
increased capacity.
• VSet phone controllers with full-color LCD displays and Telos® Status Symbols present producers and
talent with a rich graphical information display. Each VSet features its own address book and call log.
• Drop-in modules can integrate VX phone control directly into your Axia mixing consoles.
• Included XScreen Lite screening software with built-in soft-phone allows a “phone” connection on any
networked PC. Integrated recorder/editor simplifies recording of off-air conversations.
• Clear, clean caller audio from fifth-generation Telos Adaptive Hybrid technology, including Digital
Dynamic EQ, AGC, adjustable caller ducking, and send- and receive-audio dynamics processing by
Omnia®.
• Wideband acoustic echo cancellation from Fraunhofer IIS completely eliminates open-speaker
feedback.
• Support for G.722 codec enables high-fidelity phone calls from SIP clients.
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TELOS | VX BROADCAST VOIP
IN DEPTH
VoIP for Broadcast. From Telos, Naturally.
VX is the world’s first VoIP (Voice over IP) talkshow system. It’s incredibly powerful, very flexible, and
highly scalable — a powerful whole-plant broadcast phone system that’s also economical enough for
stations with just two or three studios.
VoIP has already taken the business world by storm, increasing the flexibility of office phone systems
and PBXs while simultaneously lowering maintenance and equipment costs. In fact, most Fortune 500
companies have replaced their older PBX systems with VoIP for just these reasons. There’s no reason
broadcasters shouldn’t take advantage of this cost-saving technology as well.
Sure, VX can connect to traditional POTS and ISDN telephone lines using standard Telco gateways. But
we’ve built VX around the VoIP standard, so that it can connect natively to VoIP-based PBX systems
and modern SIP Trunking services, allowing you to take advantage of low-cost Internet-delivered phone
services. In addition to cost savings from digital phone service provisioning, VX significantly eases the
cost of installation, maintenance and cabling by using standard Ethernet as its data backbone.
As a result VX is naturally scalable, capable of serving even the largest of facilities — while remaining
surprisingly cost-effective for even single stations with more modest needs. There are major operational
benefits as well. VX combines the flexibility and economy of modern SIP networking with powerful
digital signal and audio processing — making it easier than ever for talent to take control of their phone
system. You can move and share lines between studios at the touch of a button. VX is truly the future of
broadcast phones.
Why VoIP for Broadcast?
VoIP is a natural for broadcasters. Using VoIP, you can interconnect the phone system CPU with audio
interfaces, phone sets, console controllers, and PCs running screening software using efficient, low-cost
Ethernet. You can finally share phone lines among multiple studios and route caller audio anywhere in
your facility, easily and instantly. Got a hot talkshow that suddenly needs more lines in a certain studio?
Just a few keystrokes at a computer and you’re ready — no delays, and no cables to pull. VX can even
connect with your business office’s VoIP PBX to facilitate easy call transfers.
Of course, it’s got to sound good. And it does, thanks to more than two decades of DSP hybrid
technology developed by Telos. Every incoming line has its own fifth-generation digital hybrid, our most
advanced ever, packed full of technology engineered to extract the cleanest, clearest caller audio from
any phone line — even noisy cellular calls. Multiple lines can be conferenced with superior clarity and
fidelity. Smart AGC ensures consistent caller audio levels. New Acoustic Echo Cancellation from FhG
removes feedback and echo in open-speaker studio situations. And if you choose to use SIP Trunking
telco services, calls from mobile handsets with SIP clients will benefit from VX’s native support of the
G.722 codec, instantly improving caller speech quality.
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TELOS | VX BROADCAST VOIP
Since VX uses Ethernet as its network backbone, it naturally plugs right into Axia IP-Audio networks,
connecting multiple channels of audio and control using a single Ethernet cable. If you don’t have an IP-Audio
network yet, that’s OK; Telos Alliance xNodes provide AES audio and GPIO connections that work with
your existing studio equipment.
VX Components
The VX Engine
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, consoleintegrated controllers, etc.
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TELOS | VX BROADCAST VOIP
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 (Realtime Transport Protocol).
The Coolest Broadcast Phone Controllers Ever
With decades of experience designing broadcast phone systems, it’s no wonder broadcasters agree
that Telos makes the industry’s most powerful, most flexible system controllers. All VSet phones can
be powered by PoE from a Telos-approved switch, a PoE port on an Axia console engine, or by using the
included power injector.
VSet12
The VSet12 phone controller is an IP-based phone set with two large, high-contrast color LCD panels
that provide line status and caller information. VSet phones can work like a traditional Telos controller,
with calls being selected, held, and dropped in the way to which operators have grown accustomed.
But because the VX system is so powerful, much more functionality is unlocked: You can now spread
multiple calls over a number of faders, using one for each call so that operators can control each line’s
level individually. You can hard-assign individual lines to fixed faders, such as for VIP calls. You can even
map groups of lines to a single fader.
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VSet6
VSet6 is a six-line phone controller for VX. Like the VSet12, it has a bright, attractive LCD color display
with Status Symbols that feed talent instant information about line and caller status, and controls
that enable talent to step through queued calls, busy incoming lines, lock calls on-air, start an external
recording device, et cetera. Next Call functionality speeds workflow for producers, screeners, and talent.
With all the control functions of the VSet12, it’s great for smaller or secondary studios.
On-Console Control
Live calls or pre-recorded, interviews or audience participation, one thing’s certain: Phone segments are
an integral part of today’s fast-paced radio. But up to now, the phone system was separate from the onair console; audio was shared, but little else. Wouldn’t it be great if talent could take control of phones
without ever having to divert their attention from the board?
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They can: IP-Audio networking technology provides the ideal way to integrate broadcast phones into the
on-air console — the control center of every studio. VX connects directly to Axia mixing consoles using
Livewire IP-Audio to eliminate the cost and complexity of old-style inputs, outputs, and mix-minuses.
Multiple phone lines – each with a dedicated hybrid – can automatically map to individual console faders
for complete control of caller audio. And users enjoy seamless console integration, with phone controls
right on the board so that talent can dial, answer, screen, and drop calls without ever diverting their
attention from the console. Information about line and caller status can be displayed right on the console
as well.
There are plenty of other advantages to melding phones with consoles. Like ease of installation: IPAudio consoles with built-in phone controllers don’t need any additional wires or connections. Their
control signaling, caller audio, and backfeeds ride on the network connection that’s already there.
Bringing caller audio into the IP-Audio domain makes it routable like any other audio source. With the
Virtual Mixers built into Axia consoles, you could even choose to dynamically conference multiple lines
and control their gain with a single fader. And since the console now communicates directly with the
phone hybrid, mundane tasks such as mix-minus generation, starting recording devices, and playback of
recorded off-air conversations can all be automated.
Audio Interfaces
Telos Alliance xNodes let you connect VX to any non-networked radio console or other broadcast
equipment, using standard AES/EBU interfaces. A GPIO Logic xNode provides control logic where
needed. To cover all your bases, the Telos Alliance Mixed Signal xNode provides one mic/line analog input
(switchable); two analog line inputs (dedicated); three analog line outputs; one AES3 input, one AES3
output, and two GPIO ports, each with five opto-isolated ins and outs.
The Telos Alliance AES/EBU audio xNode 4 AES/EBU inputs and 4 AES/EBU outputs. Left and right input
signals may be split and routed independently as mono signals. Stunning performance specs include 48
kHz sampling rate, 126dB of dynamic range, and <0.0003% THD.
Each Telos Alliance GPIO logic xNode interface provides six general-purpose logic ports each with five
opto-isolated inputs and five outputs. A logic port can be associated with any audio input or output and
routes control data transparently along with the audio.
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VSet Call Controller
Want a VX system, but don’t have an Axia mixing console? No problem—Telos
provides VSet Console Controller electronics packages, which may be fitted to your
console using panels supplied by your OEM console provider or preferred third-party
fabricator. Like the VSet12 phoneset, the VSet Console Controller provides visual linestatus indicators and fast-take keys for selection and control of up to 12 callers,
along with standard controls such as Take, Drop, Hold and Busy keys, and the
Telos-exclusive “Next Call” key to speed workflow for producers, screeners, and
talent. There’s also a built-in keypad for on-console dialing of outgoing numbers.
Broadcast Bionics XScreen Call-Screening Software Included
XScreen software comes with every VX Engine purchase and provides call control, call screening, data
capture, and chat functionality enabling you to quickly answer, screen, and route calls using multiple
PC clients. The cloud-based database keeps a log of calls and provides further alert and directory
functionality.
XScreen can record and manage caller audio (Livewire systems only) and can additionally act as a
softphone for talking to and screening callers directly through a USB headset or soundcard on your
XScreen client PC.
XScreen is available in free (Lite) and full (subscription) versions. When you install XScreen for the first
time you will receive a 90-day free trial license for the full version. After 90 days you can continue to use
the full version with an annual subscription, or use the reduced, Lite functionality free of charge. Please
download your XScreen software from www.xscreen2.com.
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TELOS | VX BROADCAST VOIP
SPECIFICATIONS
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. (Higherquality 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 system-wide
• 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
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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, 20 Hz to 20 kHz
• Analog Line Input CMRR: >60 dB, 20 Hz to 20 kHz
VX Engine
IP/Ethernet Connections
• One 100/1000BASE-T Ethernet via RJ-45 LAN connection
• One 100/1000BASE-T 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)
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Power Supply AC Input
• Modular, field-replaceable 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+™ AES67 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.711μ-Law and A-Law, and g.722.
• Control: standard SIP trunking
Regulatory
North America: FCC and CE tested and compliant, power supply is UL approved.
Europe: Complies with the European Union Directive 2002/95/EC on the restriction of the use of certain
hazardous substances in electrical and electronic equipment (RoHS), as amended by Commission
Decisions 2005/618/EC, 2005/717/ EC, 2005/747/EC (RoHS Directive), and WEEE.
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TELOS | HX6
Hx6 Six-Line POTS Talkshow System
Give Your Phones an Instant Upgrade
OVERVIEW
Hx6 is our most advanced six-line digital Talkshow system. It features two high-performance digital
hybrids and includes our famous Digital Dynamic EQ, noise gate, caller ducking, and acoustic echo
cancellation. Works with POTS analog phone lines. Single-cable Ethernet hookup via Axia® Livewire® I/O,
or choice of analog or AES/EBU I/O with one input and one output per hybrid, and one Program On-Hold
input. Includes complimentary XScreen call-screening software from Broadcast Bionics.
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FEATURES
• Six-line capacity; works with POTS (analog) phone lines.
• Our most advanced digital hybrids, with DSP algorithms optimized for superior performance with
today’s wide variety of far-end call types (VoIP, cell, POTS, app-based).
• Telos® DDEQ (Digital Dynamic EQ) and adjustable smart-level AGC ensure spectrally consistent audio
from call to call — even on notoriously tough cellular calls.
• Excellent trans-hybrid loss of >55dB.
• Smooth, proven, symmetrical wide-range AGC by the audio processing experts at Omnia®.
• Studio adaptation and a subtle, inaudible pitch shifter to prevent feedback in open-speaker studio
environments.
• A sophisticated caller override that improves performance and allows precision adjustment of the
degree to which talent audio “ducks” the caller audio.
• Striking Telos VSet6 six-line phone controllers with large, colorful VGA LCD displays that provide
intuitive operation and setup. Telos-exclusive Status Symbols provide producers and talent with
animated, high-contrast icons that communicate line and caller status at a glance.
• Caller ID displayed on the VSet6 phoneset and the included XScreen call-screening application.
• Livewire IP-Audio allows fast, one-cable integration with Axia networks, and provides Axia board
operators with seamless, on-console control of multiple lines and hybrids. Standard Ethernet
backbone provides a common transport path for both studio audio and telecom needs, resulting in
cost savings and a simplified studio infrastructure.
• Choice of standard Analog I/O or optional, extra-cost AES/EBU I/O.
• Easy setup and configuration via Ethernet using any PC and your favorite Web browser.
• XScreen call-screening software from Broadcast Bionics, provided at no cost.
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TELOS | HX6
IN DEPTH
Advanced caller management and superior sound
Say hello to Hx6, the most advanced six-line broadcast phone system Telos has ever made. Thanks
to its Telos DSP hybrids and a full suite of audio processing capabilities, an Hx6 in your studio is like
an instant audio upgrade for on-air phone calls — song requests, morning show phoners, or callintensive talk shows.
Hx6 works with POTS phone lines, and comes equipped with two advanced telephone hybrids (each with
its own independent AGC, noise gate, and caller override dynamics) for high-quality conferencing — the
same advanced DSP technology used in the best-selling Telos Hx1 and Hx2 telephone hybrids.
The DSP toolkit in Hx6 is full-featured, to say the least. Telos Digital Dynamic EQ, our renowned adaptive
3-band processor, analyzes and adjusts received audio spectral characteristics so that calls sound
smooth and consistent despite today’s wide variety of phone sets and connection types. Adjustable
Omnia smart-level AGC with noise gating provides spectrally consistent audio from call to call — even
on notoriously tough cellular calls. A sophisticated caller override allows precision adjustment of the
degree to which talent audio “ducks” the caller audio, and exclusive feedback reduction functions help
eliminate open-speaker howl.
Like all Telos talkshow systems, the Hx6 front panel is simple and informative, with separate send and
receive meters for each hybrid, a Program-On-Hold audio presence indicator, a high-resolution OLED
display for setup, and navigation keys for quick adjustments.
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TELOS | HX6
Around back, you’ll find audio I/O, GPIO, and Telco connections. Hx6 connects directly to 6 POTS lines.
Separate analog or optional AES digital I/O is provided for each hybrid, as well as a Program-On-Hold
input, GPIO connections for speaker muting, ring tallies, et cetera.
There’s also an Ethernet port. This provides connection of as many as six Telos VSet phones, but that’s
not all: It’s also an Axia Livewire port. Through that jack, Hx6 puts audio, hybrid control, and mix-minus
for all six phone lines onto one single skinny CAT-5 cable. Livewire setup is simple: Plug it into your
Axia network, do some fast web-based configuration, and your talent can control Hx6 directly from an
Axia mixing console equipped with Call Controller modules. The Ethernet connection also allows for
convenient remote setup and administration.
With all of these capabilities, you’d expect Hx6 to cost twice as much — but it doesn’t. In fact, you can
have an Hx6 for about what you’d pay for some other companies’ “premium” systems.
Intuitive, easy-to-use controllers
This is the Telos VSet6 six-line phone controller, an IP-based phoneset with a large, high-contrast
color LCD panel that provides line status and caller information. There’s almost no learning curve; VSet
phones work like traditional Telos controllers, with calls selected, held, and dropped in the way to which
operators have grown accustomed. Exclusive animated Telos Status Symbol icons show line and caller
status at a glance; easy VSet controls let talent manage incoming lines, lock calls on-air, start an external
recording device, and take a queue of calls to air sequentially, for precise management of multi-call
interviews or conferences. Next Call functionality speeds workflow for producers, screeners, and talent.
The LCD display delivers detailed line status, caller information, caller ID, time ringing-in or on-hold, and
even comments entered in the included Xscreen screening software. A built-in address book and call
history log round out VSet6’s features. And, just like the Hx6 itself, each VSet6 has its own web server
for easy remote configuration and software upgrades.
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TELOS | HX6
Axia On-Console Control
Hx6 works with any brand of broadcast console. But wouldn’t it be great if talent could take control of
phones without ever having to divert their attention from the board? Whether your shows consist of
live calls or pre-recorded interviews, phone segments are usually fast-paced with little room for error.
But traditionally, the phone system was separate from the on-air console, making it hard to use both
together efficiently, leading engineers and talent to ask: “Why can’t the console and the phone system
work together?”
Now, they can. Hx6 can connect directly to Axia mixing consoles using Livewire IP-Audio to eliminate
the cost and complexity of old-style inputs, outputs, and mix-minuses. IP-Audio networking technology
provides the ideal way to integrate broadcast phones into the on-air console — the control center of
every studio. Users enjoy seamless console integration, with phone controls right on the board so that
talent can dial, answer, screen, and drop calls without ever diverting their attention from the console.
Information about line and caller status can be displayed right on the console as well.
There are plenty of other advantages to melding phones with consoles. Like ease of installation: IPAudio consoles with built-in phone controllers don’t need any additional wires or connections. Their
control signaling, caller audio, and backfeeds ride on the network connection that’s already there.
Bringing caller audio into the IP-Audio domain makes it routable like any other audio source. And
since the console now communicates directly with the phone hybrid, mundane tasks such as mixminus generation, starting recording devices, and playback of recorded off-air conversations can all be
automated. All of which means faster, more precise phone segments — since operators’ eyes never
need to leave the console.
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TELOS | HX6
Broadcast Bionics XScreen Call-Screening Software Included
XScreen software comes with every Hx6 Engine purchase and provides call control, call screening,
data capture, and chat functionality enabling you to quickly answer, screen, and route calls using
multiple PC clients. The cloud-based database keeps a log of calls and provides further alert and
directory functionality.
XScreen can record and manage caller audio (Livewire systems only) and can additionally act as a
softphone for talking to and screening callers directly through a USB headset or soundcard on your
XScreen client PC.
XScreen is available in free (Lite) and full (subscription) versions. When you install XScreen for the first
time you will receive a 90-day free trial license for the full version. After 90 days you can continue to use
the full version with an annual subscription, or use the reduced, Lite functionality free of charge. Please
download your XScreen software from www.xscreen2.com.
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TELOS | HX6
SPECIFICATIONS
General
• Telos 3rd-generation Adaptive Digital Hybrids
• Telos Exclusive Feedback Reduction Functions
• Send-to-Caller Processing: High-pass Filter, Frequency Shifter, AGC/Limiter, Program-on-Hold AGC/
Limiter, Sample Rate Conversion (with AES option)
• Receive-From-Caller Processing: High-pass “Hum” Filter, Smart AGC / Platform Leveler, Noise
Gate, Telos DDEQ (Digital Dynamic Equalization) 3-band Adaptive Spectral Processor, Sample Rate
Conversion (with AES option)
Analog Inputs
• Send Analog Inputs: 2x
• Program-on-Hold Analog Inputs: 1x
• Connector: XLR Female, Pin 2 High (Active Balanced with Protection)
• Input Level: Adjustable from -7 to +8 dBu (nominal)
• Analog Clip Point: +21 dBu
• Impedance: Bridging, > 10K Ohms
• Analog-to-Digital Converter Resolution: 20 bits
Analog Outputs
• Receive Analog Outputs: 2x
• Connector: XLR Male, Pin 3 High
• Output Level: Adjustable from -7 to +8 dBu (nominal)
• Impedance: <50 ohms
• Digital-to-Analog Converter Resolution: 24 bits
• Headroom Before Clipping: 20 dB headroom above 4dBU nominal levels
Switching Matrix and Conferencing
• Audio Routing and Switch: All Digital
• Telephone Lines: 6
• Hybrids: 2
• Studio Inputs: 2
• Studio Outputs: 2
• Program-on-Hold: 1
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TELOS | HX6
Control Ports
Ethernet 100BASE-T
• Web server for configuration and software update
• Telnet for command line control and diagnostics
• Call Screening Interface server allows up to 8 instances of call-screening software to connect
simultaneously
General purpose Input/Output
• 2x 15-pin D-sub with status outputs and control inputs
Control Interface
• Up to 12 attached controllers (any mix of VSet6 phones,
Console Controllers or screening software) via Ethernet connection
Power Supply
• Type: Internal auto-ranging, 85–250 VAC auto-switching, 50–60 Hz
• Power consumption: 14.2 Watts
Analog Telephone Connectivity
• Universal interface for worldwide application
• Programmable loop current
• Programmable ring and disconnect signaling (loop drop or tone)
• Programmable Flash time
• Caller ID decoding using Bellcore 212 modem standard
Regulatory
North America: FCC and CE tested and compliant, power supply is UL approved.
Europe: Complies with the European Union Directive 2002/95/EC on the restriction of the use of certain
hazardous substances in electrical and electronic equipment (RoHS), as amended by Commission
Decisions 2005/618/EC, 2005/717/ EC, 2005/747/EC (RoHS Directive), and WEEE.
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TELOS | IQ6
iQ6 Six-Line POTS Telco Gateway
Talkshow System for Axia® IP-Audio Networks
OVERVIEW
Telos® iQ6 is a six-line digital phone system designed specifically for use with Axia networked mixing
consoles — that’s why we call it a “Telco gateway.” iQ6 acts as a portal for Axia systems, supplying caller
audio, mix-minus, Program-On-Hold audio, and switching control for six POTS phone lines, using a single
RJ-45 network connection.
iQ6 is built around the Telos Hx6, our most advanced six-line POTS Talkshow system. It features two
high-performance digital hybrids and includes Telos’ famous Digital Dynamic EQ, noise gate, caller
ducking, and acoustic echo cancellation. Includes complimentary XScreen call-screening software from
Broadcast Bionics.
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TELOS | IQ6
FEATURES
• Single-cable Ethernet connection to Axia IP-Audio networks transports caller audio, mix-minus,
Program-On-Hold audio and hybrid switching control – no separate audio connections or contact
closures to solder.
• Direct, on-console control of iQ6 operations, with add-on modules for popular Axia iQ, Element® and
Axia Fusion® mixing consoles. Talent never needs to take their eyes off the control board; shows run
smoother with less errors.
• Works with Telos VSet6 six-line phone controllers with large, colorful VGA LCD displays that provide
intuitive operation and setup.
• Telos-exclusive Status Symbols on-console and phone controllers provide producers and talent with
animated, high-contrast icons that communicate line and caller status at a glance.
• Six-line capacity; works with POTS (analog) phone lines.
• Our most advanced digital hybrids, with DSP algorithms optimized for superior performance with
today’s wide variety of far-end call types (VoIP, cell, POTS, app-based).
• Telos DDEQ (Digital Dynamic EQ) and adjustable smart-level AGC ensure spectrally consistent audio
from call to call — even on notoriously tough cellular calls.
• Excellent trans-hybrid loss of >55dB.
• Smooth, proven, symmetrical wide-range AGC by the audio processing experts at Omnia®.
• Studio adaptation and a subtle, inaudible pitch shifter to prevent feedback in open-speaker
studio environments.
• A sophisticated caller override that improves performance and allows precision adjustment of the
degree to which talent audio “ducks” the caller audio.
• iQ6 versions matched to your choice of analog POTS phone lines.
• Caller ID.
• Easy setup and configuration via Ethernet using any PC and your favorite Web browser.
• XScreen call-screening software from Broadcast Bionics, provided at no cost.
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TELOS | IQ6
IN DEPTH
Six lines of crystal-clear caller audio.
One easy RJ-45 connection.
A multi-line phone system that connects to your console with just one cable? Smooth, detailed caller
audio — even from cellular callers? That’s iQ6, the no-hassle Telco gateway for Axia mixing consoles.
iQ6 plugs right into Livewire® AoIP network networks, saving money and time by eliminating the cost
and labor of old-fashioned discrete I/O, cabling, and soldered connectors. All connections to and from the
iQ6 system — receive and send audio, hybrid control, mix-minus for six phone lines, even connections
to VSet6 phone controllers and included PC-based call-screening software — travel over a single skinny
CAT-5 cable. Setup is simple: Plug it into your Axia network, do some fast web-based configuration, and
voila! you’re taking calls.
The photo above shows a complete Axia iQ console system, with QOR.32 console engine, iQ6 Talkshow
system, and iQ control surface with onboard phone controller. Control of both iQ6 hybrids and Status
Symbols information icons are right on the mixer’s surface.
You can also pair iQ6 with Telos Vset phones and their full-color, high-contrast display screens. iQ6
is extremely flexible: You can connect up to 12 control devices at once — phones, PCs, or console
controllers — to take charge from nearly anywhere. Separate Send and Receive level meters for each
hybrid are conveniently located right on the front panel for extra monitoring confidence.
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TELOS | IQ6
How does iQ6 sound? Like a Telos, of course! Inside, two of our most advanced hybrids handle up to
six POTS phone lines. Those hybrids are equipped with Digital Dynamic EQ and adjustable smart-level,
symmetrical wide-range AGC by Omnia to keep callers sounding clean, clear, and spectrally consistent
call after call. An adjustable caller override lets you dial-in just the right amount of call ducking. Our
subtle, inaudible pitch-shifter helps prevent open-speaker feedback. And conference linking lets you set
up high-quality conferencing between callers at the touch of a button — no external equipment needed.
The iQ6 front panel is simple and informative, with separate send and receive meters for each hybrid,
a Program-On-Hold audio presence indicator, a high-resolution OLED display for setup, and navigation
keys for quick front-panel adjustments.
The back panel likely looks much different from any other phone system you’ve seen. There are no
discrete audio I/O, GPIO, PoH or output connections — the RJ45 connection to your Axia Livewire
network handles all of that. Like its brother, the Telos Hx6, iQ6 connects directly to 6 POTS lines.
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TELOS | IQ6
On-Console Control
iQ6 connects directly to Axia mixing consoles using Livewire IP-Audio to eliminate the cost and
complexity of old-style inputs, outputs, and mix-minuses. IP-Audio networking technology provides
the ideal way to integrate broadcast phones into the on-air console. Users enjoy seamless console
integration, with phone controls right on the board so that talent can dial, answer, screen, and drop calls
without ever diverting their attention from the console. Information about line and caller status can be
displayed right on the console as well, with Telos Status Symbols icons that communicate line and caller
status at a glance — ensuring that phone segments are always smooth and error-free.
There are plenty of other advantages to melding phones with consoles. Like ease of installation: IPAudio consoles with built-in phone controllers don’t need any additional wires or connections. Their
control signaling, caller audio and backfeeds ride on the network connection that’s already there.
Bringing caller audio into the IP-Audio domain makes it routable like any other audio source. And
since the console now communicates directly with the phone hybrid, mundane tasks such as mixminus generation, starting recording devices, and playback of recorded off-air conversations can all be
automated. All of which means faster, more precise phone segments — since operators’ eyes never
need to leave the console.
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TELOS | IQ6
VSet6 Phone Controllers
For off-console control, iQ6 works with functional, easy-to-use Telos VSet6 six-line phone controllers.
Their big, colorful VGA LCD displays with animated high-contrast Status Symbols make fast work of
call screening, queue placement and other tasks; built-in controls for profanity delay and record devices
round out its useful toolset.
VSet6 is an IP-based phoneset that also connects to Axia networks with a single Ethernet connection.
There’s almost no learning curve; VSet phones work like traditional Telos controllers, with calls selected,
held, and dropped in the way to which operators have grown accustomed. Easy VSet controls let talent
manage incoming lines, lock calls on-air, start an external recording device, and take a queue of calls to
air sequentially, for precise management of multi-call interviews or conferences. Next Call functionality
speeds workflow for producers, screeners, and talent. The LCD display delivers detailed line status, caller
information, caller ID, time ringing-in or on-hold, and even comments entered in the included Xscreen
screening software. A built-in address book and call history log round out VSet6’s features. Each VSet6
has its own web server for easy remote configuration and software upgrades.
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TELOS | IQ6
Broadcast Bionics XScreen Call-Screening Software Included
XScreen software comes with every iQ6 Engine purchase and provides call control, call screening, data
capture, and chat functionality enabling you to quickly answer, screen, and route calls using multiple
PC clients. The cloud-based database keeps a log of calls and provides further alert and directory
functionality.
XScreen can record and manage caller audio (Livewire systems only) and can additionally act as a
softphone for talking to and screening callers directly through a USB headset or soundcard on your
XScreen client PC.
XScreen is available in free (Lite) and full (subscription) versions. When you install XScreen for the first
time you will receive a 90-day free trial license for the full version. After 90 days you can continue to
use the full version with an annual subscription, or use the reduced, Lite functionality free of charge.
Please download your XScreen software from www.xscreen2.com.
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TELOS | IQ6
SPECIFICATIONS
General
• Telos 3rd-generation Adaptive Digital Hybrids
• Telos Exclusive Feedback Reduction Functions
• Send-to-Caller Processing: High-Pass Filter, Frequency Shifter, AGC/Limiter, Program-on-Hold AGC/
Limiter, Sample Rate Conversion (with AES option)
• Receive-From-Caller Processing: High-pass “Hum” Filter, Smart AGC / Platform Leveler, Noise
Gate, Telos DDEQ (Digital Dynamic Equalization) 3-band Adaptive Spectral Processor, Sample Rate
Conversion (with AES option)
Input / Output Channels
• Send channels: 2x
• Receive channels: 2x
• Program-on-Hold channels: 1x
• Connection: 100-BaseT Ethernet (Livewire)
Switching Matrix and Conferencing
• Audio Routing and Switch: All Digital
• Telephone Lines: 6
• Hybrids: 2
Control Ports
• Ethernet 100BASE-T
• Web server for configuration and software update
• Call-Screening Interface server allows up to 8 instances of call screening software to connect
simultaneously
• GPIO channels: 2x
• Control Interface: Up to 12 attached controllers (any mix of VSet6 phones, Console Controllers or
screening software) via Ethernet connection
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TELOS | IQ6
Power Supply
• Type: Internal auto-ranging, 85–250 VAC auto-switching, 50–60 Hz
• Power consumption: 14.2 Watts
Analog Telephone Connectivity
• Universal interface for worldwide application
• Programmable loop current
• Programmable ring and disconnect signaling (loop drop or tone)
• Programmable Flash time
• Caller ID decoding using Bellcore 212 modem standard
Regulatory
North America: FCC and CE tested and compliant, power supply is UL approved.
Europe: Complies with the European Union Directive 2002/95/EC on the restriction of the use of certain
hazardous substances in electrical and electronic equipment (RoHS), as amended by Commission
Decisions 2005/618/EC, 2005/717/ EC, 2005/747/EC (RoHS Directive), and WEEE.
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TELOS | HX1 & HX2
Hx1 & Hx2 Digital Hybrids
POTS phones never sounded so good.
OVERVIEW
Telos® Hx1 one-line and Hx2 two-line POTS telephone hybrids are the most advanced hybrids ever
developed for use with analog phone lines. Hx hybrids contain advanced 3rd-generation Telos hybrids
for superior audio quality; universal POTS interface features disconnect-signal detection, which works
with Telco providers worldwide. Hx hybrids include unique features to make operators’ lives easier,
such as Auto-Answer with selectable ring count, a switchable mic/line input, call screening and linehold features, and front-panel send and receive audio metering. Audio sweetening tools include
Telos Digital Dynamic EQ (DDEQ) and adjustable smart leveler, symmetrical wide-range AGC and
noise gating by Omnia®, studio adaption and pitch shifter for use in open-speaker applications, and
adjustable caller override.
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TELOS | HX1 & HX2
FEATURES
• Single (Hx1) or two-line (Hx2) capacity with standard analog I/O (1 each send and receive in/out for
Hx1, 2 each for Hx2).
• Convenient switchable mic/line input.
• AES/EBU digital audio I/O option available at time of order, or as a field upgrade kit.
• Our most advanced digital POTS hybrids ever, with DSP algorithms optimized for superior performance
with today’s wide variety of incoming call types.
• Front-panel send and receive audio metering.
• Telos DDEQ (Digital Dynamic EQ) and adjustable smart-level AGC ensure spectrally consistent audio
from call to call — even on notoriously tough cellular callers.
• Excellent trans-hybrid loss of >55dB.
• Smooth, proven, symmetrical wide-range AGC by the audio processing experts at Omnia Audio.
• Studio adaptation and a subtle, inaudible pitch shifter to prevent feedback in open-speaker studio
environments.
• Precision adjustable caller override.
• Analog I/O 24-bit A/D/A sample rate conversion, 20 dB headroom from +4 dBu nominal levels.
• AES/EBU I/O sample rate converters accept 32, 44.1, and 48 KHz rates. Clock for outputs may be
sourced from the AES inputs or internally-generated at 48 KHz.
• Incredible dynamic range of > 92 db (analog in to analog out, studio loop mode,
10 hz – 20 khz A-weighted)
• Auto-Answer with selectable ring count and disconnect-signal detection.
• Call screening and line-hold features.
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TELOS | HX1 & HX2
IN DEPTH
Take total control of your talk shows and call-in segments.
In the mid-1980s, Telos pioneered the very first digital adaptive telephone hybrid. Since then, our POTS
phone hybrids have earned a worldwide reputation for extracting clean, clear caller audio from even the
most difficult calls.
We’ve contributed plenty of improvements to POTS hybrid technology in the past 20 years, and the Telos
Hx1 and Hx2 represent the highest state-of-the-art in hybrid performance. Advances in DSP have been
pretty great as well. We’ve used every bit of knowledge gained to make Hx1 and Hx2 the best, most
advanced POTS hybrids we’ve ever made, without much doubt.
Inside the single-hybrid Hx1 and dual-hybrid Hx2, you’ll find Telos processing technologies that take
the POTS hybrid to a new level of consistently superior performance, regardless of telephone line
characteristics. This advanced hybrid technology brings new standard features that sweeten and control
caller audio better than ever before; features you won’t find in other POTS hybrids.
On the front panel, you’ll find EQ Meters for each hybrid, to tell you exactly how much DDEQ is being
applied. Next to those, separate Send and Receive level meters monitor each hybrid. There’s also an
animated line status display that visually indicates when a line is ringing in, on air, on hold or available. A
complement of Take, Hold and Drop buttons complete the front-panel control set.
Around back (Hx2 rear panel shown above), you’ll find a switchable mic/line input, balanced analog
receive-out output, RJ ports for telco input and phoneset, input level adjustment, and a DB9 remote
control connector with GPIO closures for hybrid control and status indicator lamps. Need digital I/O? No
problem — Hx comes in an AES/EBU version with built-in sample-rate converter.
Hx1 and Hx2 are probably the most fully-featured POTS hybrids ever created, with Auto-Answer,
caller disconnect detection, audio-leveling and anti-feedback routines for open-speaker applications,
call screening and line-hold features, and much, much more. Audio processing tools include a new
symmetrical wide-range AGC and noise gate by Omnia, with adjustable gain settings to help keep
caller audio smooth and consistent from call to call. Adjustable caller override improves performance
even further, and allows you to individualize the degree to which the announcer ducks the caller audio.
Finally, our famous Digital Dynamic EQ, coupled with an adjustable smart leveler, keeps audio spectrally
consistent from call to call.
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TELOS | HX1 & HX2
SPECIFICATIONS
General
• Telos 3rd-generation Adaptive Digital Hybrid.
• Telos Exclusive Feedback Reduction Functions.
• Send-to-Caller Processing: High-pass Filter, Frequency Shifter, AGC/Limiter, Sample Rate Conversion
(with AES option).
• Receive-From-Caller Processing: High-pass “Hum” Filter, Smart AGC / Platform Leveler, Noise
Gate, Telos DDEQ (Digital Dynamic Equalization) 3-band Adaptive Spectral Processor, Sample Rate
Conversion (with AES option)
Analog Inputs:
• Send Analog Inputs: 1 for Hx1, 2 for Hx2 (one per hybrid)
• Connector : XLR Female, Pin 2 High (Active Balanced with RF Protection)
• Input Range: Selectable between MIC and LINE levels
• Line Input Level: Adjustable from -10dBV to +8 dBu (nominal)
• Analog Clip Point : +21 dBu
• Impedance: Bridging, > 50 Ohms
• Analog-to-Digital Converter Resolution: 24 bits
Analog Outputs:
• Receive Analog Outputs: 1 for Hx1, 2 for Hx2 (one per hybrid)
• Connector: XLR Male, Pin 3 High (Active Balanced, RF suppressed
• Output Level: Nominal +4 dBu, fixed
• Impedance: < 50 Ohms
• Digital-to-Analog Converter Resolution: 24 bits
• Headroom Before Clipping: 20 dB headroom above 4 dBU nominal levels
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TELOS | HX1 & HX2
AES3 Digital Inputs/Outputs (optional)
Plug-in module converts standard analog XLR inputs and outputs to AES3 (one input or output on left
channel of AES stream)
• Conforms to AES3 standard
• Sample rates: 32kHz to 48kHz.
• Rate conversion: Input and output, independently selectable
• Output Clock: AES input or 48kHz internal.
• Input Level: Nominal at -20 dBFs.
• Output Level: Nominal at -20 dBFs
Audio Performance
• Frequency Response: 200 to 3400 Hz, +/- 1 dB
• THD+N: < 0.5% THD+N using 1 KHz sine wave
• Dynamic Range: Analog in to Analog out, studio loop mode, 10Hz-20Khz. A-weighted: > 92 dB
• SNR: Analog output, referred to -12dBm phone line signal (+4dBu studio out), 10Hz-20Khz
a-weighted: > 72 dB
• Trans-Hybrid Loss: Analog phone line with ducking, gate, AGC, EQ
• all OFF relative to +4dBu input level: >55 dB
Analog Telephone Line Connectivity
• Universal interface for worldwide application. Programmable loop current, ring signaling, and flash
time. Includes caller ID decoding using Bellcore 212 modem standard.
Power Supply
• Type: Internal auto-ranging, 90–265 VAC auto-switching, 50–60 Hz.
• Power consumption: 100 Watts.
Regulatory
North America: FCC and CE tested and compliant, power supply is UL approved.
Europe: Complies with the European Union Directive 2002/95/EC on the restriction of the use of certain
hazardous substances in electrical and electronic equipment (RoHS), as amended by Commission
Decisions 2005/618/EC, 2005/717/ EC, 2005/747/EC (RoHS Directive), and WEEE.
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OMNIA | OMNIA.11
Omnia.11 FM and FM/HD
Top-of-the-line maximum firepower for an
extremely competitive environment.
OVERVIEW
Omnia.11 is available in FM+HD with separate processing paths for FM and HD/DRM or FM without
HD/DRM. The FM-only model is upgradeable to FM/HD at a later date. Switchable Single Sideband
Suppressed Carrier (SSBSC) technology for potential reduction of multipath is a standard feature. A front
panel touch screen GUI, on a 10.5” diagonal screen, provides ease of use and enhanced metering and
diagnostics. Remote access is available via any web browser. Livewire, AES/EBU digital and analog I/O
are standard. Fanless cooling. Rugged 4 RU chassis.
AUDIO PROCESSING | FM | FM+HD | AM | MULTICASTING | CODED AUDIO | STUDIO APPLICATIONS
TELOSALLIANCE.COM
OMNIA | OMNIA.11
Omnia.11s now ship with the G-Force™ Dynamics Engine standard. G-Force is also available as an
optional Plug-In for Omnia.11 units already in the field. Designed by Frank Foti and Cornelius Gould,
Omnia.11 with G-Force represents a significant update to Omnia.11’s dynamics—so significant, in
fact, that Omnia has updated the GUI to a vivid cobalt blue. It sounds even better than it looks thanks
to a dynamics processing framework that enables the Omnia.11 to set the overall EQ for signature
consistency, making it sound cleaner, clearer, louder, more consistent, more open, and more pleasing.
You hear the music. You hear the voice. You don’t hear the processor. Both FM+HD and FM-only models
can be upgraded with the optional Perfect Declipper Plug-In, a revolutionary new algorithm engineered
by audio-processing legend Hans van Zutphen that restores clipped areas in audio recordings. This
algorithm not only restores dynamics, but removes distortion.
FEATURES
G-Force™ Dynamics Engine
The G-Force Plug-In (which ships standard on all new Omnia.11 units and can be added as an optional
upgrade for existing units in the field) lets Omnia.11 to handle rapidly changing, hyper-compressed
source material better than ever with new, sophisticated improvements. The G-Force dynamics
processing framework enables the Omnia.11 set the overall EQ for signature consistency, making it
sound cleaner, clearer, louder, more consistent, more open, and more pleasing. You hear the music. You
hear the voice. You don’t hear the processor.
Solar Plexus
For deep, tight bass that you can feel!
1 Louder
To gain that extra db of loudness.
Intelligent Ultra-Multiband Limiter System
Self-adjusting attack/release functions guarantee crystal clear music and voice. The limiters are selfadapting and can tune themselves to the activity of the AGC section on the fly, providing more powerful
and transparent limiter action than possible before. Makes adjustment of limiters a breeze!
Bass Management
Manages harmonics for a natural and undistorted bottom end.
Ultra LoIMD Distortion Controlled Clipper System
Dramatically reduces intermodulation distortion (IMD) for more loudness headroom.
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MPX Composite Baseband Over AES (Omnia Direct)
Output of the Omnia.11’s stereo generator can be coupled directly to the modulator of the transmitter’s
exciter. This enables the exciter to modulate with more precision and clarity.
Perfect Declipper Plug-In (Optional)
A revolutionary new algorithm to restore clipped segments and remove distortion in aggressively
mastered audio recordings results in a clearer, more open texture that also gives more flexibly with
processing choices. (Must be running v3.0 and G-Force.)
SSBSC Technology
Omnia.11 Single Sideband Suppressed Carrier (SSBSC) technology may reduce multipath distortion.
Extra Wide Touchscreen
10.5” diagonal screen clearly shows all controls.
Looks Cool and Stays Cool
Military-grade, fanless industrial design stays cool with heatsinks in rugged chassis.
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IN DEPTH
New G-Force™ Dynamics Engine
G-Force (which ships standard on new units and is available as an optional Plug-In to upgrade existing
field units) has a highly refined density detection scheme, which means rock-solid performance across a
wide range of recordings. Program adaptive attack, release, and ratio values let you set the characteristic
elements of your signature sound and make audio acceleration and deceleration smoother than
ever. A Makeup Threshold allows for gain management and control without sudden, audible swings.
Additionally, AGC sections synchronize with program material. Multiband Limiters now self-adapt to the
Multiband AGC activity and also feature program-controlled attack and release, actively reducing limiterinduced inter-mod distortion. Limiters are more responsive and active, yet remarkably transparent, even
under extreme activity. G-Force requires v3.0, outlined below.
Version 3.0
The G-Force plug-in runs on Omnia.11 Version 3.0 system software. Included in this general system
release are many improvements, including: Static RDS; the flexibility of analog, AES/EBU or Livewire
patch points; patch point location for PPM® chains so you can integrate your Voltair/Encoder combo
into your audio processing chain; patch point input and output meters for easy level references; upper
subharmonic control of Solar Plexus over tonal balance; and compatibility with the Day-Sequerra
Time Lock system. A parametric EQ including lo and hi shelf replaces the limiter mixer section in the
HD processing chain, adding compatibility with the new patch point location for more flexibility. You
can select whether the HD processing is fed with the signal that goes through the patch point or not,
allowing for separate PPM® watermarking of HD/digital sides. Finally, all input and output “Master” level
controls now adjust in 0.1dB steps, with an “accelerator” mode giving 0.5dB steps. V3.0 also lets you
demo the G-Force Plug-In with no obligation.
Omnia.11 v3.0 is available as a complimentary, downloadable field update at TelosAlliance.com/omnia/
omnia11. Any Omnia.11 running version 1.5 or higher can perform this update.
The Perfect Declipper Plug-In Option
Just when you thought nothing could sound any better, G-Force-enabled Omnia.11s can be upgraded
with the Perfect Declipper Plug-In. Engineered by audio-processing legend Hans van Zutphen, the
Perfect Declipper uses a revolutionary new algorithm to replace clipped areas in audio recordings,
restoring dynamics and removing distortion. You must be running v3.0 and G-Force to install the Perfect
Declipper Plug-In.
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Intelligent Ultra-Multiband Limiter System
Traditional limiting technology has often resulted in various forms of audio corruption. Omnia.11’s new
LoIMD technology coupled with smart gain reduction algorithms now have limiters that sound amazingly
transparent. All AGC and limiting algorithms employ an auto acceleration / deceleration mechanism,
which tunes out perceptible intermodulation distortion. The attack/release functions adjust themselves
based upon content density. This breakthrough method literally analyzes the audio content in both
the amplitude and frequency domain, then adapts the timing networks—on the fly—to transparently
control the signal, without the control being heard. The result is revealed in added detail, clarity, and
quality, yet maintaining the desired competitive loudness level. Special attention was paid to the
behavior of live voice quality. The improved performance of the AGC and limiter functions generate live
voice clarity and impact far beyond that which was previously possible.
Bass Management
The bass enhancement algorithm is a key feature of the Omnia.11. Low end is now broadcast with
recording studio-like punch and impact, with no traditional side-effects whatsoever.
Omnia.11’s exclusive bass-management method is a mixture of innovation, as well as a rearrangement
of the system topology. Achieving great-sounding bass requires the most effort, partly due to the fact
that the bass spectrum has the most number of harmonics, and all of these must be kept properly
accounted for in the time domain. Also, any additional spectra created (enhancement) must have its
harmonic content managed, or the bass region begins to sound distorted and unnatural. This process
requires much more than just traditional EQ, bass clipping/filtering, or any ordinary attempt at bass
enhancement. Even the location where the function is inserted matters, as well as how it maintains
its frequency range along with the rest of the system. An entire dissertation could be done on the
bass enhancement/management system alone. The classic Omnia dynamically flat & time aligned
crossover system has been further refined to produce smooth, rich, and full tonality. The AGC and
limiter sections cannot be fooled into false gain control due to spectral density (or lack thereof) from
the crossover network.
G-Force takes this all to another level of greatness that allows broadcasters to adjust the Omnia.11’s
bass via a single knob. Advanced adjustment mode allows more precise bass sculpting, including
Omnia’s Solar Plexus bass enhancement feature. An intelligent active bass clipper system allows the full
power of the new bass enhancement scheme to come through on the dial.
Ultra LoIMD Distortion Controlled Clipper System
Audio processing for conventional broadcast (FM and AM) has, in some applications, reached extreme
levels. Various methods are available today capable of creating LOUD competitive signals, but at the
expense of perceptible quality. Through critical listening, extensive research, and evaluation of processing
methods, it has been determined the single most annoying quotient is due to intermodulation distortion
(IMD) induced by aggressive functions within the processing system. The algorithms are pushed to the
limits, and beyond. One of the most crucial, aggressively used algorithms in the FM processor is the
pre-emphasized final limiter/clipper. Omnia Engineering has developed the new Ultra LoIMD Distortion
controlled clipper system specifically to reduce IMD in this critical stage of the processing.
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For those who feel the need to use it, there’s also a composite clipper embedded in the stereo generator.
However, to date, all of our testing has been done without any composite clipping. Pilot protection is
on the order of magnitude close to 90 dB, which is considerably more protection than necessary for
even the best FM receiver. Integrated laboratory-grade stereo generator with dual MPX outputs, 19
kHz reference output for external RDS/RBDS systems and pilot protection that provides >80dB pilot
protection—with or without composite clipping. MPX spectral low-pass filter protects RDS/RBDS and
SCA signals if composite clipping is employed. There are multiple ways to adjust the system to achieve
the exact sound you’re looking for.
Unprecedented Access
• A front panel touch screen GUI, on a 10.5” diagonal screen, provides ease of use and enhanced
metering and diagnostics. Remote access is via any web browser.
• Livewire, AES/EBU digital and analog I/O is standard. Headphone soft “patch points” are available for
listening through the processing chain.
• Diversity-Delay.
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SPECIFICATIONS
General
• Non-linear Crosstalk: > -80 dB, main to sub or sub to main channel (referenced to 100% modulation).
• 38 kHz Suppression: > 70 dB (referenced to 100% modulation).
• 76 kHz Suppression: > 80 dB (referenced to 100% modulation).
• Pilot Protection: > -65 dB relative to 9% pilot injection, ± 1 kHz.
• 57 kHz (RDS/RBDS) Protection: better than -50 dB.
• Connectors: Two EMI suppressed female BNC, floating over chassis ground
• Maximum Load Capacitance: 5nF (at 10 ohms source impedance).
• Maximum cable length: 100 feet/30 meters RG-58A/U.
Analog Audio Input
• Left/Right Stereo. Electronically balanced.
• Input impedance 10k ohms resistive.
• Maximum Input Level: +22 dBu.
• Nominal Input Level: +4dBu, which nets a -18dBFS input meterreading on a steady-state signal when
the Input Gain controlis set to 0.0dB. Program material with a nominal average level(VU reading) of
+4dBu will typically produce peak readings on the input meter in the range of -12 dBFS to -6dBFS.
This is the correct operating level.
A/D Conversion
• Crystal Semiconductor CS5361, 24 bit 128x over-sampled delta-sigma converter with linear-phase
anti-aliasing filter. Pre-ADCanti-alias filter, with high-pass filter at <10 Hz.
• Connectors: Two, EMI-suppressed XLR-female. Pin 1 chassis ground, Pin 2 “Hot.”
Analog Audio Output
• Left/Right Stereo. Electronically balanced.
• Output Impedance 20 ohms.
• Minimum load Impedance: 600 ohms.
• Output Level adjustable from -2 dBu to +22dBu peak in 0.1dB steps.
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D/A Conversion
• Crystal Semiconductor CS4391, 24 bit, 128x oversampled.
• Connectors: Two, EMI-suppressed XLR-male. Pin 1 chassis ground, Pin 2 “Hot.”
Frequency Response
• Complies with the standard 50 or 75 microsecond pre-emphasis curve within ± 0.5 dB, 30 Hz to
15 kHz. The analog left/right output and AES/EBU Digital outputs can be configured for flat or preemphasized output.
System Distortion
• Less than 0.01% THD, 20 Hz – 7.5 kHz. Second harmonic distortion above 7.5 kHz is not audible in the
FM system.
• Signal-Noise Ratio: > -80 dB de-emphasized, 20 Hz – -15 kHz bandwidth, referenced to 100%
modulation.
• The measured noise floor will depend upon the settings of the Input and Output Gain controls and is
primarily governed by dynamic range of the Crystal Semiconductor CS5361 A/D Converter which is
specified as >110 dB. The dynamic range of the internal digital signal processing chain is >144 dB.
Stereo Separation
• Greater than 65 dB, 20 Hz – -15 kHz; 70 dB typical.
Crosstalk
• > -70 dB, 20 Hz -- 15 kHz.
System Latency
• 36ms. “FM” channel, as measured from the analog inputs through the composite MPX output.
Composite Outputs
• Source Impedance: 5 ohms or 75 ohms, jumper-selectable. Single ended and floating over chassis
ground. Output Level: 0V to 10V in 0.05V steps, software adjustable.
D/A Conversion
• Texas Instruments/Burr Brown PCM1798, 24-bit sigma-delta converter.
Configuration
• Two electrically independent outputs. Software based level adjustment.
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Load Impedance
• 50 ohms or greater load is suggested.
Pilot Level
• Adjustable from 4.0% to 12.0% in 0.1% steps and OFF.
Pilot Stability
• 19 kHz, ± 0.5 Hz.
Signal-to-Noise Ratio
• -85 dB typical, 75 μs de-emphasized, 15 kHz bandwidth, referenced to 100% modulation).
Distortion
• < 0.02% THD 20 Hz – 15 kHz bandwidth, 75 μs de-emphasized, referenced to 100% modulation.
• Stereo Separation: > 65 dB, 30 Hz – 15 kHz.
• Linear Crosstalk: > -80 dB, main to sub or sub to main channel, referenced to 100% modulation.
Connector
• XLR-female, EMI-suppressed. Pin 1 chassis ground, Pin 2-3 transformer isolated, balanced, and
floating. Standard AES3 specified balanced 110 ohm input impedance.
External Sync Range
• Automatically accepts sample rates between 32kHz and 96kHz. Connector: XLR-female, EMIsuppressed. Pin 1 chassis ground, pins 2 and 3 transformer isolated, balanced, and floating – AES3
standard 110 ohm impedance.
Remote Control
• Via Ethernet using built-in Java (TM) based remote control program integrated into web page interface.
All software is served from the built-in web server to any standard web browser; there is nothing to
install on the user’s computer.
Connectors
• Ethernet - Industry standard EMI-suppressed RJ-45 connector.
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GPI Interface
• Connector: EMI suppressed DB-15 female connector.
Power Requirements
• Voltage: 100-250 VAC, 47-63 Hz.
Power Connector
• EMI suppressed IEC male. Detachable 3-wire power cords supplied for US and European use.
Power Supply
• Internal. Overvoltage and short circuit protected.
Digital Audio Input
• Configuration: Stereo per AES/EBU standard, CS8420 Digital Audio Transceiver with 24 bit resolution,
software selection of stereo, mono from left, mono from right or mono from sum.
• Automatically accepts and locks to input sample rates between 30 and 108 kHz.
• Connector: XLR-female, EMI-suppressed. Pin 1 chassis ground, pins 2 and 3 transformer isolated,
balanced, and floating – AES3 standard 110 ohm impedance.
Digital Audio Output #1
• Stereo per AES3 standard. Output can be configured in software for flat or pre-emphasized response
at 50 or 75 microseconds.
• Digital Sample Rates: Output sample rates software selectable for 48kHz, Sync to Input or Sync to
External.
• Connector: XLR-male, EMI-suppressed. Pin 1 chassis ground, pins 2 and 3 transformer isolated,
balanced, and floating. Standard AES3 specified 110 ohm source impedance.
• Digital Output Level: -22.0 to 0.0 dBFS software adjustable.
Digital Audio Output #2:
• Stereo per AES3 standard. Output can be configured in software for flat pre-emphasized response at
50 or 75 microseconds.
• Digital Sample Rates: Output sample rates software selectable for 48kHz, 44.1kHz or Sync to External.
• Connector: XLR-male, EMI-suppressed. Pin 1 chassis ground, pins 2 and 3 transformer isolated,
balanced, and floating. Standard AES3 specified 110 ohm source impedance.
• Digital Output Level: -22.0 to 0.0 dBFS software adjustable.
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External Sync Input:
• External Sync: Output sample rate can be synchronized to the signal present on the AES/EBU input, or
to an AES3 signal applied to the Ext. Sync input connector. (Does not accept Word Clock Inputs)
Regulatory
North America: FCC and CE tested and compliant, power supply is UL approved.
Europe: Complies with the European Union Directive 2002/95/EC on the restriction of the use of certain
hazardous substances in electrical and electronic equipment (RoHS), as amended by Commission
Decisions 2005/618/EC, 2005/717/ EC, 2005/747/EC (RoHS Directive), and WEEE.
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Omnia.9
Top-of-the-line FM or AM processing. Optional
HD-1, HD-2, HD-3 processing and streaming/
encoding, RDS. Also available in Dual Path model.
OVERVIEW
FM or AM processing on demand is standard. Simultaneous FM+AM if programming is 100% simulcast
and at same transmitter site. Studio output with very low latency for talent monitoring is also standard.
Optional HD-1, HD-2, HD-3 processing and streaming/encoding, RDS. Dual path model available for two
separate program feeds broadcasting on FM+FM, FM+AM or FM+FM+AM simulcast with HD output,
IP-Streaming with separate processing and encoding and low latency studio processing, all built-in for
both feeds.
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FEATURES
Exclusive “Undo” Technology
“Undo” is a two-step process that restores peaks and dynamic range to - and removes distortion from source material that has been damaged by over-compression and clipping during the mastering process.
Psychoacoustically Controlled Distortion-Masking Clipper
Omnia.9’s final FM clipper takes into account how the human ear hears and perceives distortion and
uses that information to effectively mask it, leaving only clean, distortion-free audio on the air.
Omnia® Toolbox
Every Omnia.9 comes with a “toolbox” that includes loudness metering, a digital oscilloscope, an FFT
spectrum analyzer, and real time analyzer (RTA) to help you adjust your processing and “see what you’re
hearing”.
Speaker Calibration
With the addition of a calibrated microphone, Omnia.9’s built-in real time analyzer (RTA), pink noise
generator, and parametric equalizer can be used to calibrate any speaker system.
Dry Voice Detector
The dry voice detector is a selectable feature that helps eliminate the audible distortion sometimes
evident on bare voice (voice with no music mixed under it) when very aggressive processing settings
are used.
Remote Client
Omnia.9’s client software allows full remote control of the processor from any Windows-based PC or
tablet, including touch screen devices, on the local network. The remote interface looks and functions
just like the front panel screen.
Auto Pilot
A selectable feature that automatically turns off the 19kHz stereo pilot when Omnia.9 detects that the
source material is mono, and turns it back on when the input is determined to be stereo, automatically
gaining a 20 dB increase in signal to noise during mono programming.
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Additional Features
• Multiband downward expansion (source noise reduction)
• 3-stage wideband AGC with adjustable sidechain equalization
• Program-dependent multiband compression
• Multiband look-ahead limiting
• Selectable phase linear high pass filter, 15, 30 or 45 Hz
• Two-band final look-ahead limiting on digital paths
• 7 inch front panel touch screen
• Full remote control via included NFRemote software with client audio streaming for remote
adjustment and monitoring. Also supports full remote control and monitoring via HTTP request
• On-screen keyboard with several layouts (QWERTY, QWERTZ, AZERTY, Dvorak and ABC sequential) for
easy setup and preset name typing
• Selectable SSB (Single Sideband) stereo encoder
• HTTP push support for automation, such as RDS and streaming song titles, preset recall
• Studio Output with very low latency for talent monitoring
• Dual independent power supplies
• Composite pass-through (relay bypass) for your backup processor, with full analysis tools for external
composite signals
• Internal patch point capability so you can optimize placement of your Voltair / PPM® encoder within
your audio processing chain.
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IN DEPTH
“Undo”
The first step of Undo is the de-clipper, which examines and recreates audio peaks that were clipped
during mastering. The second step is a multi-band expander that creates dynamic range. Clean,
well-recorded audio has always been able to withstand greater degrees of processing. This was true
decades ago and it’s still true (and more relevant than ever) today. An FM processor, by its very nature,
compresses dynamic range and employs some form of clipping to deliver a “signature sound” and a
competitively loud signal on the air. It is an unfortunate but well-accepted fact that recordings made
in the past two decades have been on the decline in terms of quality, as mastering engineers seem to
be waging their very own “loudness wars”. The result is source material that is hyper-compressed right
out of the jewel case with only a dB or two of dynamic range at most. As if that weren’t bad enough,
the music is run through unsophisticated, brute-force clippers to make them louder still. The result is
that the audio going INTO a processor today sounds more distorted than the audio coming OUT of an
FM air chain 10 years ago! Before it even gets touched by the compressors, limiters, and clippers in the
processor itself, it has been damaged. Rip a track from the modern CD of your choice and look at the
waveform in your favorite editor if you need proof. Processors add more distortion still, and the resulting
“music” heard on the air is nearly unlistenable. By repairing the damaged audio first, “Undo” gives
Omnia.9 cleaner and more dynamic audio to work with, which can better stand up to the rigors of on-air
processing. The result is a clean, dynamic, and listenable sound on the air. In fact, audio processed by
Omnia.9 for FM often sounds far better than the original CD.
Psychoacoustically Controlled Distortion-Masking Clipper
Clipping is typically the final stage of an FM processing chain. The majority of clipping is usually done in
the final L/R audio, with additional, optional clipping available in the composite signal. The final clipper is
also where the classic (and oft dreaded) “loud v. clean” tradeoff is made. When more clipping is used to
gain loudness on the dial, clipper distortion becomes more and more pronounced. The clipped peaks fall
back into the audio and manifest themselves as audible distortion.
There are ways to get around that problem, but they come at a price. You could back down on the clipper
drive to clean up the sound, but then you lose loudness. Or, you could put more of the “heavy lifting” on
the compressors and limiters preceding the clipper, but that results in an overly busy, dense sound that
robs the music of life and causes listener fatigue. Some processors HAVE to resort to building excess
density in the dynamics section because their simple or old-technology clippers simply aren’t up to the
job. The Omnia.9 identifies clipper distortion and uses a proprietary psychoacoustic-controlled algorithm
in the composite signal to mask it, effectively eliminating it from the final audio. It is so robust that it
boasts an additional 3dB of high-frequency headroom and is capable of 140% L/R modulation within
100% total modulation. That means Omnia.9 can be significantly cleaner for a given loudness level or
substantially louder for a given level of quality. It comes closer to eliminating the “either/or” compromise
than any other processor on the market today.
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Omnia Toolbox
When Leif Claesson was creating Omnia.9, he knew that having diagnostic and measurement tools
would be necessary. The original plan was to keep them in place only for development, but he quickly
realized that engineers would find great value in them as well, and decided to leave them in place.
Audio processing is largely a “hearing” process, but there is much to be learned by seeing what your
adjustments are doing to your sound as well. Some stations still have an oscilloscope on the test bench
or a spectrum analyzer at the transmitter, but it’s not always convenient (or possible) to hook up a
processor to them while it’s on the air.
Even if you did so, you’re pretty much limited to monitoring only the composite output of your own
station’s processing. Omnia.9’s built-in solution means there’s no extra test equipment to buy (‘scopes
and analyzers aren’t cheap) and no cables to hook up. It also means you can visually monitor the signal
at the input, the output, and dozens of other points throughout the processing path so you can tell
what’s happening to the audio every step of the way. As an added bonus, Omnia.9’s composite inputs
can be fed from a calibrated tuner or frequency-agile mod monitor so that you can monitor the other
signals in your market too!
In addition to these tools, Omnia.9 also includes RTA and speaker calibration tools to further assist with
monitoring and fine-tuning your processing. While it is certainly good practice to listen to your station
on a variety of radios and speakers as you adjust your processing, it is also good practice to have at
least one set of calibrated speakers available. Otherwise, the changes you make to your processing
will be influenced by listening to speakers that don’t accurately reflect the frequency response of your
processing adjustments. By adding an inexpensive calibrated microphone and using the included pink
noise generator and RTA, you can quickly and easily calibrate a set of speakers to use as a reference as
you adjust your sound.
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Speaker Calibration
If you make decisions about your processing on uncalibrated monitors, you are making choices that
are influenced by the differences in frequency response present in every speaker, not to mention the
coloration imposed by the room in which you are listening. Simply put, you’re dealing with subjective,
not objective, information. By using the pink noise generator and RTA built into Omnia.9 and adding
an inexpensive calibrated microphone, it is possible to calibrate any speaker system to deliver as flat a
response as the speakers themselves will allow (small speakers still won’t reproduce low frequencies
as well as larger ones – the laws of physics still apply after all). With speaker and room influences
removed from the equation, you are in a position to adjust your audio based only upon “the facts.”
When explaining this process to someone in person, this is the point in the conversation where they
inevitably say, “But listeners aren’t hearing my station on calibrated speakers! They’re listening in their
cars, at their computer, and through cheap ear buds, so I should too!” It’s true - that’s exactly how your
listeners are hearing your station in the real world, and why it is always important to listen on a wide
variety of radios in many different environments. But adjusting your processing this way is a shortcut
to a lot of tail-chasing frustration and lousy audio. Let’s say you listen first in an inexpensive compact
car with a typical factory stereo. You notice there isn’t much bass, so you adjust your processing to
deliver more low end. It sounds good. Then you move into a high-end luxury car with 10 speakers and a
subwoofer, and the bass is muddy, boomy, and overwhelming. Why? Because you adjusted the bass in
the processor to make up for deficiencies you thought were in your processing, but in fact were in your
speakers! Having at least one pair of high quality, calibrated speakers to go back to as your reference
will dramatically improve your on-air sound, save you valuable time, and help preserve your sanity at
the same time. (Don’t worry – there are still plenty of people at your station to chip away at your mental
well-being – we just don’t want to be among them!).
Dry Voice Detector
We know that the human voice can present a tough challenge to an FM processor. If it’s bare voice – that
is, voice alone with no music mixed underneath – any distortion created in the processing really stands
out. We also know that all-out loudness comes at a price: At some point, you have to give up “clean”
to get “loud.” Even Omnia.9’s psychoacoustically-controlled distortion-masking clipper, which really
minimizes the dreaded “clean v. loud” tradeoff, can reveal some distortion on dry voice when the overall
processing is set up to really push for loudness. So ensure clean voice quality in these situations, the dry
voice detector first determines that the incoming audio is actually bare voice. It then automatically and
inaudibly transfers more of the “heavy lifting” to the compressor and limiter sections, thereby reducing
the amount of overall clipping needed to maintain the same level of loudness.
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Remote Client
Every modern processor provides some means by which to control it or adjust its settings remotely,
which is handy if the processor is at a transmitter site miles (and often mountains) away from the studio.
Most employ web-based interfaces, which on the surface sounds convenient because it allows you to
remote in from a browser on any computer at any location, but even the best of them fall short when it
comes to a great user experience. They require browser plug-ins, typically feel “laggy” when viewing
meters or adjusting controls, and don’t always have the same look and feel as the front panel interface.
Omnia.9’s client software delivers exactly the same experience whether you’re standing in front of the
processor or controlling it from your PC or tablet. If you have Omnia.9’s on more than one station in your
group (who can buy just one?) you can connect to any of them through a single connection window, and
can run multiple remotes simultaneously.
Providing your connection has sufficient bandwidth, you can even stream audio from various patch
points within the processing chain back to the client computer. This allows you to hear what effect your
adjustments have on your audio in the environment of your choice instead of a rack room or transmitter
building, locations which almost never have decent monitors but offer noise in abundance!
Auto Pilot
The ability to transmit 2-channel source audio in FM stereo certainly has a sonic advantage, but it’s
far from a “free ride”. The stereo pilot typically claims around 9% of total modulation; stereo signals are
more susceptible to noise and multipath distortion than mono FM signals; and for a given RF level at
the receiver, the signal-to-noise ratio of a stereo signal will be worse than that of a mono signal. Those
are acceptable tradeoffs if you’re actually playing stereo music, but if your source material is mono (be
it mono music or talk programming), it hardly seems fair to force those compromises upon your signal
when there’s no reason to do so.
Thus, the Auto Pilot feature will automatically turn off the pilot (resulting in mono transmission) when
the audio is mono, lowering the noise floor by 20dB and stopping multipath in its tracks when it is
most audible.
AUDIO PROCESSING | FM | FM+HD | AM | MULTICASTING | CODED AUDIO | STUDIO APPLICATIONS
TELOSALLIANCE.COM
OMNIA | OMNIA.9
SPECIFICATIONS
Frequency Response
• +/-0.5dB 20Hz to 15kHz, 17.5kHz in extended mode
Signal to Noise Ratio
• Greater than -80dBu de-emphasized, 20Hz to 15kHz
System Distortion
• Less than 0.01% THD below pre-emphasis, inaudible above
Stereo Separation
• 65dB minimum, 20Hz to 15kHz, 70dB typical
Digital Output Level
• Adjustable from -24.0dBFS to 0.0dBFS in 0.1dB increments
Stereo Baseband Output
• Adjustable from -2dBU to +22dBU (0.1dB increments) into 600-Ohms, 20-Ohm output impedance
A/D Conversion
• Crystal Semiconductor CS5361, 24 bit 128x over-sampled delta sigma converter with
linear-phase anti-aliasing filter.
• Pre-ADC anti-alias filter, with high-pass filter at <10 Hz
D/A Conversion
• Crystal Semiconductor CS4391, 24-bit, 128x oversampled
External Sync Input
• Per AES11 Digital Audio Reference Signal (DARS), reference for digital output sample rate.
Range is 32kHz to 96kHz.
Analog I/O
• Two balanced, EMI filtered XLR connectors
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OMNIA | OMNIA.9
Stereo Generator Connections
• Four 75-Ohm BNC female, two inputs, two outputs
• (FM style only) AES/EBU In & External Sync
Digital I/O
• AES/EBU via four XLR connectors for Main and Aux Digital programs (two stereo in, two stereo out)
Ethernet
• Shared RJ45 supporting 100 and 1000BASE-T Ethernet connections
Power Requirements
• 100-264 VAC, 47-63Hz autosensing, dual PSU
Power Connector
• Dual IEC male, detachable 3-wire power cords supplied
Power Supply
• Internal dual redundant, hot-swappable
Environmental
• Operating: 0 to 50 degrees C
• Non-operating: –20 to 70 degrees C.
Regulatory
North America: FCC and CE tested and compliant, power supply is UL approved.
Europe: Complies with the European Union Directive 2002/95/EC on the restriction of the use of certain
hazardous substances in electrical and electronic equipment (RoHS), as amended by Commission
Decisions 2005/618/EC, 2005/717/ EC, 2005/747/EC (RoHS Directive), and WEEE.
AUDIO PROCESSING | FM | FM+HD | AM | MULTICASTING | CODED AUDIO | STUDIO APPLICATIONS
TELOSALLIANCE.COM
OMNIA | OMNIA.9SG
Omnia.9sg
For “Split” audio processing applications
OVERVIEW
Omnia.9 Technology in a 1RU Standalone Stereo Generator
In some applications, it might make more sense to locate the main audio processor at the studios
while the stereo generator and limiter live out at the transmitter site. This might include cases where
the STL uses discrete analog or AES audio rather than composite, or situations where one primary
audio processor feeds multiple sites such as satellite distribution.
Omnia.9sg brings the same acclaimed clipper, limiter, and stereo generator technology in the Omnia.9
to a standalone box—but that isn’t all it does…We brought a few other tricks from the Omnia.9 along
for the ride.
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OMNIA | OMNIA.9SG
FEATURES
• Composite embedding allows up to 140% L/R modulation at
100% total modulation.
• Full IP remote control with remote audio streaming
• Optional RDS encoder
• Selectable SSB (Single Sideband) stereo encoder
• Analysis tools for internal or external composite signals
IN DEPTH
Psychoacoustically Controlled Distortion Masking Clipper
As with the Omnia.9, the psychoacoustically controlled distortion-masking clipper built into the
Omnia.9sg allows for incredibly loud audio without distortion.
ITU-BS.412 MPX Power Controller
With all of this capability for ultimate loudness and modulation, the ITU-BS.412 MPX Power Controller
ensures you remain compliant with applicable regulations while still producing competitively loud yet
clean audio.
Auto Pilot
Another feature carried over from the Omnia.9 into the 9sg is the unique “Auto Pilot” function. Turning
off the stereo pilot during mono programming allows for a 20 dB signal-to-noise improvement
when it counts the most. Auto Pilot will automatically suppress the 19 kHz stereo pilot during mono
programming ensuring that your programming stays as noise-free as possible.
Full Remote Control
While the front panel LCD allows basic configuration and monitoring, the Omnia.9sg has a complete IP
remote control interface. NF Remote software (also used with many other Telos Alliance products) can
control the unit and access the same full suite of test instrumentation available in the Omnia.9. External
composite signals from modulation monitors can also be analyzed and streamed through the remote
software for off-air monitoring.
AUDIO PROCESSING | FM | FM+HD | AM | MULTICASTING | CODED AUDIO | STUDIO APPLICATIONS
TELOSALLIANCE.COM
OMNIA | OMNIA.9SG
SPECIFICATIONS
Frequency Response:
• +/- .5 dB 20 Hz to 15 kHz
Signal-Noise Ratio:
• Greater than -80 dBu de-emphasized, 20 Hz to 15 kHz
Stereo Separation:
• 65 dB minimum, 20 Hz to 15 kHz, 70 dB typical
Stereo Baseband Output:
• Adjustable from -2 dBu to +22 dBu (0.1 dB increments) into 600-Ohms, 20-Ohm output impedance
Analog Inputs:
• Two balanced, EMI filtered XLR connectors
Digital Inputs:
• AES/EBU In & External Sync
Composite I/O:
• Four 75-Ohm BNC female, two inputs, two outputs
Remote Control:
• RJ45 supporting 100BASE-T Ethernet connections
Power Requirements:
• 100-264 VAC, 47-63Hz autosensing, dual PSU
Power Connector:
• Dual IEC male, detachable 3-wire power cords supplied
AUDIO PROCESSING | FM | FM+HD | AM | MULTICASTING | CODED AUDIO | STUDIO APPLICATIONS
TELOSALLIANCE.COM
OMNIA | OMNIA.9SG
Environmental:
• Operating: 0 to 50 degrees C
• Non-Operating: -20 to 70 degrees C
Regulatory
North America: FCC and CE tested and compliant, power supply is UL approved.
Europe: Complies with the European Union Directive 2002/95/EC on the restriction of the use of certain
hazardous substances in electrical and electronic equipment (RoHS), as amended by Commission
Decisions 2005/618/EC, 2005/717/ EC, 2005/747/EC (RoHS Directive), and WEEE.
AUDIO PROCESSING | FM | FM+HD | AM | MULTICASTING | CODED AUDIO | STUDIO APPLICATIONS
TELOSALLIANCE.COM
OMNIA | OMNIA.7AM
Omnia.7AM
AM Never Sounded This Good
OVERVIEW
Do you want your AM broadcasts to sound CLEANER, CLEARER, and LOUDER? Then you need a
processor designed and built for today’s AM radio.
Meet Omnia.7AM, a feature-rich, competitively priced AM audio processor engineered to do just that,
and the first processor dedicated to AM to appear on the scene in many years.
While competitors are using old and outdated technology in their processors, the Omnia.7AM uses the
best of current technology to meet current challenges faced by AM broadcasters. Omnia.7 AM delivers
the powerful, clear, and precise Omnia signature sound that’s the first choice of top stations worldwide.
The Omnia.7AM employs most of the features of the 7FM. All aspects of the processing infrastructure,
bandwidth, and their output signals, however, have been specially engineered for maximum efficiency
and performance within the AM spectrum.
AM Never Sounded This Good
AUDIO PROCESSING | FM | FM+HD | AM | MULTICASTING | CODED AUDIO | STUDIO APPLICATIONS
TELOSALLIANCE.COM
OMNIA | OMNIA.7AM
FEATURES
• “Undo,” exclusive Omnia technology that removes distortion and mathematically re-creates the peaks
sliced from today’s poorly mastered contemporary music. Undo restores life, brilliance, and dynamic
range to any type of music.
• An exclusive Psychoacoustic Controlled Distortion Masking Clipper analyzes and masks distortion
perceptible to the human ear, leaving only clean, clear audio.
• A complete toolbox of sophisticated Omnia sound-shaping technology gives you the power to analyze
and refine your signature sound using a variety of sonic tools ranging from Real-Time Analyzers to
Oscilloscopes, FFTs, and more.
• Remote client software allows full remote control of processor and all metering tools from any
Windows-based PC or tablet on the local network—including touch-screen devices.
• Dry Voice Detector detects speech and applies appropriate processing for clearest possible voice
quality.
• Built-in Speaker Calibration tool.
• Multiband downward expansion (source noise reduction).
• Three-stage wideband AGC with adjustable sidechain equalization.
• Program-dependent two- to five-band multiband AGCs and limiters.
• 4.3” / 10.9 cm. front panel screen.
• Full remote control with audio monitoring.
• Dual, independent power supplies.
• Composite pass-through (relay bypass) for backup processor.
• Asymmetrical output with 150% maximum positive peak modulation, 100% maximum negative peak
modulation, and peak inversion controls for the input and output audio.
• Available with optional HD and/or streaming.
AUDIO PROCESSING | FM | FM+HD | AM | MULTICASTING | CODED AUDIO | STUDIO APPLICATIONS
TELOSALLIANCE.COM
OMNIA | OMNIA.7AM
IN DEPTH
“Undo”
A processor, by its very nature, compresses dynamic range and employs some form of clipping to deliver
a “signature sound” and a competitively loud signal. Clean, well-recorded audio has always been able to
withstand greater degrees of processing; this was true decades ago and it’s more relevant than ever today.
Unfortunately, recordings made in the past two decades have declined in terms of quality, as mastering
engineers wage their very own “loudness wars.” (Rip a track from the modern CD of your choice and look
at the waveform in your favorite editor if you need proof!)
The result is source material that is hyper-compressed from the studio, with only a dB or two of dynamic
range at most. As if that weren’t bad enough, the music is run through unsophisticated, brute-force
clippers to make them louder still. Before it even gets touched by the compressors, limiters, and clippers
in the processor itself, it has been damaged.
By repairing the damaged audio before processing, “Undo” gives Omnia.7AM cleaner and more dynamic
audio to work with. The first step of Undo is the declipper, which examines and mathematically recreates audio peaks that were flattened during mastering. The second step, a multi-band expander,
increases dynamic range. The result: clean, dynamic, enjoyable sound.
Psychoacoustically Controlled Distortion-Masking Clipper
Clipping is typically the final stage of a processing chain. The final clipper is also where the classic (and
oft dreaded) “loud versus clean” tradeoff is made. When more clipping is used to gain loudness on the
dial, clipper distortion becomes more and more pronounced. The clipped peaks fall back into the audio
and manifest themselves as audible distortion.
How to solve this problem? You could back down the clipper drive to clean up the sound — but you lose
loudness. You could dial up the compressors and limiters that precede the clipper — but that results in
busy, dense sound that leads to listener fatigue.
To put it plainly: Omnia.7AM sounds significantly cleaner than other processors at a given loudness
level — and substantially louder at any given level of quality. It comes closer to eliminating the “loud /
clean” compromise than any other processor on the market today. Voices sound clean, while music and
production sounds surprisingly vibrant on AM transmission.
This is especially important for AM because unlike FM where loudness is desired for dial-dominating
sound, loudness on AM directly translates to increased signal coverage, especially on the fringes of the
signal. So it’s even more important to have the ability to be loud AND clean on AM since more and better
coverage means more potential listeners, ultimately resulting in potentially more revenue.
AUDIO PROCESSING | FM | FM+HD | AM | MULTICASTING | CODED AUDIO | STUDIO APPLICATIONS
TELOSALLIANCE.COM
OMNIA | OMNIA.7AM
Omnia Toolbox
While audio processing is largely a “hearing” process, there is still much to be learned by seeing what
your adjustments are doing to your sound. Some stations still have an oscilloscope on the test bench
or a spectrum analyzer at the transmitter, but it’s not always convenient (or possible) to hook up a
processor to them while on the air.
With Omnia.7AM, there’s no extra test equipment to buy (‘scopes and analyzers aren’t cheap) and
no cables to hook up. You also have the built-in capability to visually monitor the signal at the input,
the output, and dozens of in-between points throughout the processing path so you can tell what’s
happening to your audio every step of the way.
Speaker Calibration
If you make decisions about your processing on uncalibrated monitors, your choices are colored by the
audio characteristics of the speaker itself — not to mention those of your listening room.
The pink noise generator and RTA built into Omnia.7AM, used with an inexpensive calibrated
microphone, makes it possible to calibrate any speaker system to deliver as flat a response as the
speakers will allow. (Small speakers still won’t reproduce low frequencies well; the laws of physics still
apply!) With speaker and room influences removed from the equation, you are free to adjust your audio
based only upon “the facts.”
“But,” you say, “listeners aren’t hearing my station on calibrated speakers! They’re listening in their cars,
at their computer, and through cheap ear buds, so I should too.” It’s true — that’s exactly how your
listeners hear your station, and why listening on a variety of radios, in many different environments, is a
good idea. But adjusting your processing this way invariably results in frustration and lousy audio.
Here’s why: You listen first in a compact car with a typical factory stereo. You don’t hear much bass, so
you adjust your processing to deliver more low end. Then, you move to a luxury car with 10 speakers
and a subwoofer, and the bass is muddy, boomy, and overwhelming. Why? Because you adjusted the
processing to make up for its perceived deficiencies, when the real deficiency was in your speakers!
Having at least one pair of high-quality, calibrated speakers as your reference will dramatically improve
your on-air sound, save you valuable time—and help preserve your sanity, too!
AUDIO PROCESSING | FM | FM+HD | AM | MULTICASTING | CODED AUDIO | STUDIO APPLICATIONS
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OMNIA | OMNIA.7AM
Dry Voice Detector
The human voice can often present a real challenge. Even Omnia.7AM’s psychoacoustically controlled
distortion-masking clipper, which dramatically minimizes the dreaded “clean / loud” tradeoff, can reveal
some distortion on voices when overall loudness is the goal.
To ensure clean voice quality in these situations, the Dry Voice Detector listens for speech, then
automatically and inaudibly adjusts the compressor and limiter sections, reducing the amount of overall
clipping needed to maintain the same level of loudness.
Remote Client
Remote control is a must — especially when your processor is miles (and often mountains) away from
the studio. Omnia.7AM takes remote control to a new high, with a high-performance Web interface that
eliminates interface lag. And, if you have multiple Omnia.7AM processors, its single connection window
enables you to manage multiple remotes simultaneously.
Provided your network has sufficient bandwidth, you can even stream audio from various patch points
within the processing chain back to your computer, so you can hear the effect of your adjustments in the
quiet of your office — not inside a noisy transmitter building.
AUDIO PROCESSING | FM | FM+HD | AM | MULTICASTING | CODED AUDIO | STUDIO APPLICATIONS
TELOSALLIANCE.COM
OMNIA | OMNIA.7AM
SPECIFICATIONS
Frequency Response
• 20Hz to 10.0kHz, +/-0.5dB.
• Adjustable NRSC pre-emphasis curve implementation. You may instead choose pre-emphasis at 50us
or 75us, or even a flat output if desired. Low pass filter can be set between 3.0kHz and 10.0kHz in
0.5kHz increments. This allows maximum high-frequency transmission while allowing for stationspecific bandwidth restriction scenarios such as HD Radio, 9 kHz channel spacing, or even more
narrow-band shortwave transmission.
• +/- 0.5dB 20Hz to 15kHz; 16.5kHz in extended mode.
Signal-to-Noise Ratio
• Greater than -80 dB, de-emphasized, 20Hz to 15kHz
System Distortion
• Less than 0.01% THD+N below pre-emphasis, inaudible above
Stereo Separation
• 65dB minimum, 20Hz to 15kHz; 70dB typical
Stereo Baseband Output
• Adjustable from -24.0dBFS to 0.0dBFS in 0.1 dB increments
Digital Output Level
• Adjustable from -24.0dBFS to 0.0dBFS in 0.1dB increments
A/D Conversion
• Crystal Semiconductor CS5361, 24 bit 128x over-sampled delta sigma converter with linear-phase
anti-aliasing filter
• Pre-ADC anti-alias filter, with high-pass filter at <10 Hz
• Delta sigma converter with linear-phase and anti-aliasing filter
• MPX Inputs have high pass filter <0.1Hz
AUDIO PROCESSING | FM | FM+HD | AM | MULTICASTING | CODED AUDIO | STUDIO APPLICATIONS
TELOSALLIANCE.COM
OMNIA | OMNIA.7AM
D/A Conversion
• Crystal Semiconductor CS4391, 24-bit, 128x oversampled
• External sync input
• Per AES11 Digital Audio Reference Signal (DARS), reference for digital output sample rate
• MPX Outputs are DC coupled
Analog I/O
• Two balanced, EMI filtered XLR connectors
AES Digital I/O
• AES/EBU In; Out via XLR connectors
• Input accepts 32000 – 96000 Hz. Output is 44100 or 48000 depending on the rate selected in
software. AES Reference Input via BNC connector. Accepts 44100 or 48000 Hz only, and the correct
rate must first be selected in the software.
External Sync Range
• 44.1kHz or 48kHz
Inputs/Outputs
• Balanced, EMI-filtered, L/R analog input and output on XLR connectors
• AES input and output on XLR connectors, including recognition of external sync signal
• Ethernet RJ-45 port supporting 100 and 1000 BASE-T Ethernet. at 44.1 or 48kHz
Power Requirements
• 100-264 VAC, 47-63Hz autosensing, 100W maximum
Power Connector
• IEC male, detachable 3-wire power cords supplied
Power Supply
• Internal dual redundant
AUDIO PROCESSING | FM | FM+HD | AM | MULTICASTING | CODED AUDIO | STUDIO APPLICATIONS
TELOSALLIANCE.COM
OMNIA | OMNIA.7AM
Environmental
• Operating: 0 to 50 degrees C
• Non-operating: –20 to 70 degrees C
Physical Specifications
• Unit weight: 11 pounds
• Total shipping weight: 15 pounds
• Dimensions: 2RU at 3.5” H x 19” W x 12.5&” D
Regulatory
North America: FCC and CE tested and compliant, power supply is UL approved.
Europe: Complies with the European Union Directive 2002/95/EC on the restriction of the use of certain
hazardous substances in electrical and electronic equipment (RoHS), as amended by Commission
Decisions 2005/618/EC, 2005/717/ EC, 2005/747/EC (RoHS Directive), and WEEE.
AUDIO PROCESSING | FM | FM+HD | AM | MULTICASTING | CODED AUDIO | STUDIO APPLICATIONS
TELOSALLIANCE.COM
OMNIA | OMNIA.7FM
Omnia.7FM
Premium Performance, Priced Right.
OVERVIEW
Up to now, there have been two choices when it comes to audio processing: all the features and
advanced audio-shaping tools (with a five-figure price tag), or gear that fits your budget — but made
you compromise on performance and capabilities.
No more compromises. Meet Omnia.7FM, the premium, feature-rich FM audio processor that’s
surprisingly affordable. But low price doesn’t mean low performance: Omnia.7FM delivers the powerful,
clear and precise Omnia® signature sound that’s the first choice of top stations worldwide.
Omnia.7FM comes with a host of standard features:
• Selectable FM or HD+Streaming/Encoding
• “Undo,” exclusive Omnia technology that removes distortion and mathematically re-creates the peaks
sliced from today’s poorly mastered contemporary music. Undo restores life, brilliance, and dynamic
range to any type of music.
• An exclusive Psychoacoustically Controlled Distortion Masking Clipper analyzes and masks distortion
perceptible to the human ear, leaving only clean, clear audio.
• A complete toolbox of sophisticated Omnia sound-shaping technology gives you the power to analyze
and refine your signature sound using a variety of sonic tools ranging from loudness metering to RealTime Analyzers to Oscilloscopes, FFTs, and more.
AUDIO PROCESSING | FM | FM+HD | AM | MULTICASTING | CODED AUDIO | STUDIO APPLICATIONS
TELOSALLIANCE.COM
OMNIA | OMNIA.7FM
Simultaneous HD, Internet streaming / encoding and RDS options are also available, putting
Omnia.7FM head-and-shoulders above any other comparably priced audio processor in features,
performance, and value.
FEATURES
• Selectable FM or HD+streaming standard; optional upgrade to combinations of simultaneous
FM+HD+Streaming+RDS.
• Omnia-exclusive “Undo” Audio Restoration Technology
• Psychoacoustically Controlled Distortion Masking FM Clipper
• Two-band final look-ahead limiter for HD Radio and streaming
• Full Omnia Toolbox, with loudness metering, a digital oscilloscope, an FFT spectrum analyzer, and Real
Time Analyzer (RTA)
• Remote client software allows full remote control of processor and all metering tools from any
Windows-based PC or tablet on the local network — including touch-screen devices
• Dry Voice Detector detects speech and applies appropriate processing for clearest possible voice quality
• Built-in Speaker Calibration tool
• Multiband downward expansion (source noise reduction)
• Three-stage wideband AGC with adjustable sidechain equalization
• Program-dependent two-to-five-band multiband AGCs and limiters
• 4.3” / 10.9 cm. front panel screen
• Full remote control with audio monitoring
• HTTP push support for automation such as dynamic RDS and streaming song titles, with preset recall
• Dual, independent power supplies
• Composite pass-through (relay bypass) for backup processor
Optional Features
• Simultaneous streaming processing / encoding
• Simultaneous processing for HD Radio or DAB
• RDS encoding
AUDIO PROCESSING | FM | FM+HD | AM | MULTICASTING | CODED AUDIO | STUDIO APPLICATIONS
TELOSALLIANCE.COM
OMNIA | OMNIA.7FM
IN DEPTH
“Undo”
The first step of Undo is the de-clipper, which examines and recreates audio peaks that were clipped
during mastering. The second step is a multi-band expander that creates dynamic range. Clean,
well-recorded audio has always been able to withstand greater degrees of processing. This was true
decades ago and it’s still true (and more relevant than ever) today. An FM processor, by its very nature,
compresses dynamic range and employs some form of clipping to deliver a “signature sound” and a
competitively loud signal on the air. It is an unfortunate but well-accepted fact that recordings made in
the past two decades have been on the decline in terms of quality, as mastering engineers seem to be
waging their very own “loudness wars.”
The result is source material that is hyper-compressed right out of the jewel case with only a dB or
two of dynamic range at most. As if that weren’t bad enough, the music is run through unsophisticated,
brute-force clippers to make them louder still. The result is that the audio going INTO a processor today
sounds more distorted than the audio coming OUT of an FM air chain 10 years ago! Before it even
gets touched by the compressors, limiters, and clippers in the processor itself, it has been damaged.
Rip a track from the modern CD of your choice and look at the waveform in your favorite editor if you
need proof. Processors add more distortion still, and the resulting “music” heard on the air is nearly
unlistenable. By repairing the damaged audio first, “Undo” gives Omnia.7FM cleaner and more dynamic
audio to work with, which can better stand up to the rigors of on-air processing. The result is a clean,
dynamic, and listenable sound on the air. In fact, audio processed by Omnia.7FM often sounds far better
than the original CD.
Psychoacoustically Controlled Distortion-Masking Clipper
Clipping is typically the final stage of an FM processing chain. The majority of clipping is usually done in
the final L/R audio, with additional, optional clipping available in the composite signal. The final clipper is
also where the classic (and oft dreaded) “loud v. clean” tradeoff is made. When more clipping is used to
gain loudness on the dial, clipper distortion becomes more and more pronounced. The clipped peaks fall
back into the audio and manifest themselves as audible distortion.
There are ways to get around that problem, but they come at a price. You could back down on the clipper
drive to clean up the sound, but then you lose loudness. Or, you could put more of the “heavy lifting” on
the compressors and limiters preceding the clipper, but that results in an overly busy, dense sound that
robs the music of life and causes listener fatigue. Some processors HAVE to resort to building excess
density in the dynamics section because their simple or old-technology clippers simply aren’t up to
the job. The Omnia.7FM identifies clipper distortion and uses a proprietary psychoacoustic-controlled
algorithm in the composite signal to mask it, effectively eliminating it from the final audio. It is so robust
that it boasts an additional 3dB of high-frequency headroom and is capable of 140% L/R modulation
within 100% total modulation. That means Omnia.7FM can be significantly cleaner for a given loudness
level or substantially louder for a given level of quality. It comes closer to eliminating the “either/or”
compromise than any other processor on the market today.
AUDIO PROCESSING | FM | FM+HD | AM | MULTICASTING | CODED AUDIO | STUDIO APPLICATIONS
TELOSALLIANCE.COM
OMNIA | OMNIA.7FM
Omnia Toolbox
When Leif Claesson was first creating Omnia.9, he knew that having diagnostic and measurement tools
would be necessary. The original plan was to keep them in place only for development, but he quickly
realized that engineers would find great value in them as well, and decided to leave them in place.
The Omnia.7FM carries on the tradition!
Audio processing is largely a “hearing” process, but there is much to be learned by seeing what your
adjustments are doing to your sound as well. Some stations still have an oscilloscope on the test bench
or a spectrum analyzer at the transmitter, but it’s not always convenient (or possible) to hook up a
processor to them while it’s on the air.
Even if you did so, you’re pretty much limited to monitoring only the composite output of your own
station’s processing. Also provided as part of the Omnia.7FM, it means there’s no extra test equipment
to buy (‘scopes and analyzers aren’t cheap) and no cables to hook up. It also means you can visually
monitor the signal at the input, the output, and dozens of other points throughout the processing path
so you can tell what’s happening to the audio every step of the way. As an added bonus, Omnia.7’s
composite inputs can be fed from a calibrated tuner or frequency-agile mod monitor so that you can
monitor the other signals in your market, too!
In addition to these tools, Omnia.7FM also includes RTA and speaker calibration tools to further assist
with monitoring and fine-tuning your processing. While it is certainly good practice to listen to your
station on a variety of radios and speakers as you adjust your processing, it is also good practice to have
at least one set of calibrated speakers available. Otherwise, the changes you make to your processing
will be influenced by listening to speakers that don’t accurately reflect the frequency response of your
processing adjustments. By adding an inexpensive calibrated microphone and using the included pink
noise generator and RTA, you can quickly and easily calibrate a set of speakers to use as a reference as
you adjust your sound.
AUDIO PROCESSING | FM | FM+HD | AM | MULTICASTING | CODED AUDIO | STUDIO APPLICATIONS
TELOSALLIANCE.COM
OMNIA | OMNIA.7FM
Speaker Calibration
If you make decisions about your processing on uncalibrated monitors, you are making choices that
are influenced by the differences in frequency response present in every speaker, not to mention the
coloration imposed by the room in which you are listening. Simply put, you’re dealing with subjective,
not objective, information. By using the pink noise generator and RTA built into Omnia.7FM and adding
an inexpensive calibrated microphone, it is possible to calibrate any speaker system to deliver as flat a
response as the speakers themselves will allow (small speakers still won’t reproduce low frequencies
as well as larger ones – the laws of physics still apply after all). With speaker and room influences
removed from the equation, you are in a position to adjust your audio based only upon “the facts.” When
explaining this process to someone in person, this is the point in the conversation where they inevitably
say, “But listeners aren’t hearing my station on calibrated speakers! They’re listening in their cars,
at their computer, and through cheap ear buds, so I should too!” It’s true — that’s exactly how your
listeners are hearing your station in the real world, and why it is always important to listen on a wide
variety of radios in many different environments. But adjusting your processing this way is a shortcut
to a lot of tail-chasing frustration and lousy audio. Let’s say you listen first in an inexpensive compact
car with a typical factory stereo. You notice there isn’t much bass, so you adjust your processing to
deliver more low end. It sounds good. Then you move into a high-end luxury car with 10 speakers and a
subwoofer, and the bass is muddy, boomy, and overwhelming. Why? Because you adjusted the bass in
the processor to make up for deficiencies you thought were in your processing, but in fact were in your
speakers! Having at least one pair of high quality, calibrated speakers to go back to as your reference
will dramatically improve your on-air sound, save you valuable time, and help preserve your sanity at
the same time. (Don’t worry – there are still plenty of people at your station to chip away at your mental
well-being — we just don’t want to be among them!).
Dry Voice Detector
We know that the human voice can present a tough challenge to an FM processor. If it’s bare voice
— that is, voice alone with no music mixed underneath — any distortion created in the processing
really stands out. We also know that all-out loudness comes at a price: At some point, you have to give
up “clean” to get “loud.” Even Omnia.7FM’s psychoacoustically controlled distortion-masking clipper,
which really minimizes the dreaded “clean v. loud” tradeoff, can reveal some distortion on dry voice
when the overall processing is set up to really push for loudness. So ensure clean voice quality in these
situations, the dry voice detector first determines that the incoming audio is actually bare voice. It then
automatically and inaudibly transfers more of the “heavy lifting” to the compressor and limiter sections,
thereby reducing the amount of overall clipping needed to maintain the same level of loudness.
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Remote Client
Every modern processor provides some means by which to control it or adjust its settings remotely,
which is handy if the processor is at a transmitter site miles (and often mountains) away from the studio.
Most employ web-based interfaces, which on the surface sounds convenient because it allows you to
remote in from a browser on any computer at any location, but even the best of them fall short when
it comes to a great user experience. They require browser plug-ins, typically feel “laggy” when viewing
meters or adjusting controls, and don’t always have the same look and feel as the front panel interface.
Omnia.7FM’s client software delivers exactly the same experience whether you’re standing in front of
the processor or controlling it from your PC or tablet. If you have Omnia.7FMs on more than one station
in your group (who can buy just one?) you can connect to any of them through a single connection
window, and can run multiple remotes simultaneously.
Provided your connection has sufficient bandwidth, you can even stream audio from various patch
points within the processing chain back to the client computer. This allows you to hear what effect your
adjustments have on your audio in the environment of your choice instead of a rack room or transmitter
building, locations which almost never have decent monitors but offer noise in abundance!
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SPECIFICATIONS
Frequency Response
• +/-0.5dB 20Hz to 15kHz, 17.5kHz in extended mode
Signal to Noise Ratio
• Greater than -80dBu de-emphasized, 20Hz to 15kHz
System Distortion
• Less than 0.01% THD below pre-emphasis, inaudible above
Stereo Separation
• 65dB minimum, 20Hz to 15kHz, 70dB typical
Digital Output Level
• Adjustable from -24.0dBFS to 0.0dBFS in 0.1dB increments
Stereo Baseband Output
• Adjustable from -2dBU to +22dBU (0.1dB increments) into 600-Ohms, 20-Ohm output impedance
A/D Conversion
• Crystal Semiconductor CS5361, 24 bit 128x over-sampled delta sigma converter with linear-phase
anti-aliasing filter.
• Pre-ADC anti-alias filter, with high-pass filter at <10 Hz
D/A Conversion
• Crystal Semiconductor CS4391, 24-bit, 128x oversampled
Analog I/O
• Two balanced, EMI filtered XLR connectors
Stereo Generator Connections
• Four 75-Ohm BNC female, two inputs, two outputs
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Digital I/O
• AES/EBU In & Out via XLR connectors
• Supports stereo digital audio and Omnia Direct™
Ethernet
• Shared RJ45 supporting 100 and 1000BASE-T Ethernet connections
Power Requirements
• 100-264 VAC, 47-63Hz autosensing, dual PSU
Power Connector
• Dual IEC male, detachable 3-wire power cords supplied
Power Supply
• Internal dual redundant
Environmental
• Operating: 0 to 50 degrees C
• Non-operating: –20 to 70 degrees C.
Regulatory
North America: FCC and CE tested and compliant, power supply is UL approved.
Europe: Complies with the European Union Directive 2002/95/EC on the restriction of the use of certain
hazardous substances in electrical and electronic equipment (RoHS), as amended by Commission
Decisions 2005/618/EC, 2005/717/ EC, 2005/747/EC (RoHS Directive), and WEEE.
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OMNIA | VOLT
Omnia VOLT™
Leave the competition in the dust with audio
processing for FM, AM, SG, HD/DAB/DRM or
studio applications.
OVERVIEW
Omnia VOLT is the latest triumph from the people who brought you the multiple-award-winning
Omnia.11, the acclaimed Omnia.9, the power-packed Omnia 7, and the 13,000+ Omnia ONEs currently
in service.
But this is not just another audio processor. With VOLT, we have rewritten the rules for broadcast DSP, fine-tuning our algorithms and creating the
world’s best-sounding, most powerful, incredibly versatile 1RU audio processor. VOLT gives you more
sonic performance and processing power in one rack unit than others give you in three. We invite you to
compare this little dynamo against processors costing many thousands more.
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FEATURES
• New-Generation Frank Foti-designed Clipper. The latest most advanced thinking on clipper design
from a processing legend.
• Dynamics Engine designed by Senior Algorithm Developer Cornelius Gould.
• Six Separate AGC Sections. One wideband, plus five separate time-aligned multi-band sections, each
with separate controls for every important parameter. Plus a tunable mid-band crossover. Give your
station the loudness and consistent sound it deserves!
• Five Separate Time-Aligned Limiters, each with separate Drive, Hold, Threshold, and Attack/ Decay.
They give you protection against overmodulation while maintaining a loud signature sound.
• Variable Deep Bass, Phat Bass, and Warmth Enhancers. Get that meaty Omnia sound, fine-tuned
the way you want.
• Bass Pre-Clipper. Fully adjustable with Tightness and Girth controls. You’ll have strong, listenerpleasing bass without worrying about intermodulation distortion.
• Clipper Silk Adjustment. If your format is prone to treble distortion, you can add just enough Silk to
clean up those high frequencies.
• Sensus Processing for Digital Program streams. Omnia’s exclusive Sensus algorithms actually
predict how HD, DRM, or multicasting data reduction will affect your sound. They precondition your
signal, making compression sound better—even at low bitrates.
• Adjustable BS-412 Threshold and Processing for full compliance with ITU standards.
• Stereo Enhancement for FM Analog, without Adding Multipath. You’ll get a wider, more exciting
signal that jumps out of listeners’ radios.
• Variable High-Pass and Switchable Phase Rotator. Those ultra- low frequencies, too low to be
perceived as bass by listeners, won’t rob you of on-air power.
• Automatic Mono “Dry Voice” Sensing. Ideal for FM Analog Stereo stations using extreme processing:
it keeps an extra hand on the clipper, to stop distortion when the L+R channel gets boosted by mono
signals.
• Totally Flexible Signal Path. Use analog, AES/EBU digital, or Livewire® AoIP inputs; analog, AES/EBU
digital, Livewire, or composite outputs. Adjust channel balance and correct polarity separately on each
input. Save and recall input/output setups for different applications. All outputs are always active,
regardless of input type.
• Switchable Insert Points for Voltair®, Watermark Encoders, or Other Downstream Encoding.
Optimize your airchain and eliminate the need for external pre-processing! You can feed encoders with
a pre-processed signal from VOLT’s multiband AGC and limiters so your encoder sees a stronger, more
reliable signal. Then feed the encoder’s output back into VOLT, for post-encoding clipping that protects
you against overmodulation.
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• Automatic “Failover” signal switching. Designate a backup input to use if your main signal drops out
or STL fails. Switch to this source automatically, with adjustable sensitivity, or trigger it as needed.
• QuickTweak™ System lets you fine-tune your sound like a processing genius. Get exactly the
processing you want in minutes, while you’re on the air, right from the front panel or a connected
computer or tablet.
• Graphic User Interface is easy to navigate, but gives you the deep level of control you need.
• Built-in HTML-5 Server for full control from any computer, tablet, or smartphone… without special
plug-ins.
• Rugged 1RU Construction fits any control room, technical center, or transmitter shack, with easy-tosee LED meters.
• Cool Running, Fanless Operation. VOLT can even be used near live mics.
• Flexible Pre-Emphasis Switching makes it easy to fit VOLT into any airchain.
• Dual Variable Composite Outputs (with FM DSP|Core).
• Variable Pilot Level and Phase (with FM DSP|Core).
• Built-In Tone Generator provides for quick setup and calibration.
IN DEPTH
Nail Your Signature Sound Faster with QuickTweak™
Whether you are a processing novice or expert, Omnia VOLT gives you the tools to create a superior
signature sound. Choose from some of the best factory presets available, designed by Omnia’s
processing experts and by our favorite “insider” guest programmers. For those who want to push
beyond stock presets, Omnia’s new QuickTweak system lets you fine-tune your sound quickly. For
experts, drill down into deeper parameter adjustments. Get exactly the processing you want, while
you’re on the air, whether you’re at the front panel, or sitting in your car controlling VOLT over the web.
Nobody knows processing like Omnia. We’ve designed QuickTweak based on our decades of experience
and market leadership, algorithmically linking complex and interactive parameters to create a core set of
“meta” controls.
• QuickTweak is easy to understand: You can tune it by ear, and hear the results instantly.
• QuickTweak’s six master controls allow millions of recallable variations, right from the front panel.
• You can use QuickTweak on the factory presets, or on your own custom presets.
• You can save your own settings after using QuickTweak to easily A/B compare preset modifications.
• Any preset can be adjusted with QuickTweak, or you can fine-tune using even deeper control layers.
Presets can always be refined, then saved under a new title.
• Share presets by importing or exporting them with others in your company. Back-up preset files on
common media.
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Total Versatility with DSP|Core Firmware
VOLT’s DSP|Core firmware modules rearrange and modify VOLT as your needs change. DSP|Cores aren’t
extra cost add-ons! Download the functionality you need for free, install the DSP|Core firmware package
from a connected computer, and reboot. It’s that simple.
• Use VOLT for FM Analog Stereo at the station, with high-quality baseband clipping to feed
uncompressed STLs, or at the transmitter, with dual composite outputs.
• Use VOLT for AM Broadcast, with purpose-built presets for the challenges of AM radio. VOLT’s Tunable
Asymmetrical Modulation and Tilt controls help you get modern results, even from older transmitters!
• Use VOLT for Studio and Program Production or Syndication. It comes with the tools and presets you
need for modern production styles.
• Use VOLT for HD/DAB/DRM/Web Streaming. Our exclusive Sensus algorithms reduce compression
artifacts even at low bitrates.
• Use VOLT as a standalone FM Stereo Generator at the transmitter for direct connection to
transmitters.
• Use VOLT for low-latency FM Stereo to comply with local regulations, using a high-efficiency clipper
that’s optimized for this kind of broadcast.
Front Panel
Rear Panel
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SPECIFICATIONS
Frequency Response
• User selection of flat, 50 µs, or 75 µs pre-emphasis curve within ± 0.50 dB, 30 Hz to 15 kHz.
Signal-to-Noise Ratio
• Audio >95 dB analog, >120 dB digital I/O.
System Distortion
• Less than 0.01% THD, 20 Hz – 7.5 kHz (second harmonic distortion above 7.5 kHz is not audible in the
FM system).
Latency
• 18ms nominal, +-0.5ms depending on IO selection. Low Latency FM version <7ms
Input / Output
• Composite: Output impedance 75Ω, single-ended and floating over chassis ground. BNC connectors
with EMI suppression. Maximum cable 100’ / 30M RG-58U.
• Output level: Separately adjustable for each of two outputs, 0V - 10V in 0.05V steps.
Pilot Level: Adjustable from 4.0% to 12.0% in 0.1% steps and OFF. Pilot Stability: 19 kHz, ± 0.5 Hz. S/N:
-85 dB typical, 75 μS de-emphasized across 15 kHz, at 100% modulation Distortion: < 0.02% THD 20
Hz – 15 kHz, 75 μS de-emphasized @ 100%.
• Stereo Separation: > 65 dB, 30 Hz – 15 kHz. Linear Crosstalk: > -80 dB, main to sub or sub to main
channel @ 100%. Non-linear Crosstalk: > -80 dB, main to sub or sub to main @ 100%. 38 kHz
Suppression: > 70 dB @ 100%. 76 kHz Suppression: > 80 dB @ 100%. Pilot Protection: > -65 dB
relative to 9% pilot injection, ± 1 kHz. 57 kHz (RDS/RBDS) Protection: > -50 dB.
Analog
• Left and Right Stereo on EMI-suppressed XLR-3, balanced with “pin 2 hot.”
• Input: Electronic balanced, impedance 10kΩ, nominal +4 dBu, max +22 dBu.
• Output: Impedance 20Ω for >600Ω load, +4 dBu nominal, +22 dBu peak. Converters: 24 bit, 128x
oversampled with linear-phase anti-aliasing filter.
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Digital
• Stereo per AES/EBU standard, 24 bit resolution. Input locks to any rate 32 kHz – 108 kHz. Output
locks to input, internal 48 kHz, or separate external AES/EBU “digital black” reference 32 kHz – 96
kHz.
Audio over IP
• Audio and control over IP per Livewire standard, on same RJ-45 used for Ethernet control.
Remote Control
• GPI: EMI suppressed DB-9 at logic levels, +5 V and ground supplied. Ethernet: 10/100BaseTX.
• Ethernet on EMI-suppressed RJ-45. TCP/IP control via HTML-5 internal web server, password
protected. Manual addressing and port selection.
• Crystal Semiconductor CS5361, 24 bit 128x over-sampled delta sigma converter with linear-phase
anti-aliasing filter.
• Pre-ADC anti-alias filter, with high-pass filter at <10 Hz.
• Delta sigma converter with linear-phase and anti-aliasing filter.
• MPX Inputs have high pass filter <0.1Hz.
Electrical/Physical
• Power: 100 - 250 VAC, 47-63 Hz. < 40 VA. Typical draw 12W RMS, maximum 15W RMS. Internal
supply with overVOLTage and short circuit protection. Meets EN55022, EN55011 Level B Conducted
Emissions. EN61000-4-2, -3, -4, -5, -6 level 3 immunity compliant. Full international safety approval.
CE marked. EMI suppressed IEC male connector. Detachable 3-wire power cords supplied for US and
European use. Temperature: 32° to 122° F / 0° to 50° C for all operating VOLTage ranges.
• Humidity: 0-95% RH, non-condensing.
• Dimensions: 19” wide x 1.75” high x 16” deep (48.26cm x 13.335 cm x 40.64 cm) including connectors.
Unit requires one EIA rack space for mounting.
• Shipping Weight: 12 lbs. / 5.5 kg
Regulatory
North America: FCC and CE tested and compliant, power supply is UL approved.
Europe: Complies with the European Union Directive 2002/95/EC on the restriction of the use of certain
hazardous substances in electrical and electronic equipment (RoHS), as amended by Commission
Decisions 2005/618/EC, 2005/717/ EC, 2005/747/EC (RoHS Directive), and WEEE.
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OMNIA | OMNIA ONE
Omnia® ONE
FM, AM, HD Radio, DAB, DRM, Studio Pro, Stereo
Generator, multicasting, podcasting, netcasting
or satcasting unit as needed.
OVERVIEW
12,000+ Omnia ONEs. That’s how many are on the air around the world. Engineers tell us of reliability
and ease of operation. Program Directors love the power and punch of genuine Omnia processing.
General Managers love the price!
FEATURES
FM
Smart wideband AGC followed by advanced Four-Band AGC and selectable Four or Five-Band Peak
Limiter sections. Omnia’sadvanced, fully distortion-controlled, pre-emphasized final limiter / clipper.
A newly designed digital stereo generator with SCA convenience input, two independently adjustable
composite MPX outputs and 19kHz pilot output for synchronization to external RDS generators.
AM
Omnia’s advanced NRSC compliant, distortion-managed final limiter / clipper, including selectable Low
Pass Filter frequencies that support AM HD transmission installations...the same as used in the Omnia
ONE’s bigger siblings.
Multicast
Features SENSUS®, an audio conditioning technology to minimize codec artifacts as well as restore the
fullness and depth that bit-reduction steals. See expanded explanation about SENSUS technology later
in this brochure. Ultra low-distortion final limiting optimized for the HD codec.
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Studio Pro
Full-bandwidth processor for applications that require minimal delay and do not require absolute peak
limiting. The first studio processor to include a four-band compressor / limiter allowing you precise and
accurately defined control while pre-processing music, commercials, remote feeds, or sweetening audio.
Applications include recording studios, mastering labs, TV stations, radio headphone feeds ... just abut
any application where signal processing is needed.
Additional Features
• Price includes software downloads for FM, AM, internet/satellite, studio processing`, or Stereo
Generator on demand if necessary
• Balanced XLR Analog inputs and outputs, plus Digital AES/EBU input, output and external Sync
• Browser-based remote control and configuration
• Automatic backup input switching
• Livewire® / Ethernet RJ45 jack
• Universal Power Input
• Robust headphone amp with rugged jack and front panel volume control
• Single Jog-Wheel and Selection Button user interface with LED level metering and LCD screen
IN DEPTH
Time alignment.
The Omnia ONE is completely, 100% time-aligned. This means that all audio signals, no matter what
frequency, have the exact same propagation time from Input to Output of the Omnia audio processor. This
is a claim that other manufacturers cannot make, as they don’t deem time-alignment to be important.
Omnia processors sound more precise and less “smeary” due to this attention to time-alignment.
Pre-emphasis placement.
All other FM audio processors employ FM pre-emphasis prior to their multi-band limiting. This
placement of the pre-emphasis function results in a more convenient design for the manufacturer.
However, it becomes this approach in the processor which ends up sounding more dense and “packed
up” in the higher audio frequencies. Most engineers and listeners have come to accept this sound as “the
FM sound”. This audio aberration is not a function of FM transmission, but a bad result of the design of
traditional, multiband FM audio processors. Omnia takes a different approach. In the Omnia ONE, the
necessary FM pre-emphasis is applied after the multi-band limiting. This technique requires difficult
attention to both the limiting and clipping algorithms. However, the audible result is a cleaner, much
more detailed high-end in the transmitted audio. This advantage of Omnia processing architecture is
most easily noticed on musical instruments such as cymbals, castanets, trumpets, and other sounds
with a lot of high-frequency energy. You will notice that these musical instruments sound “fake” through
other audio processors, but they sound very real and natural through an Omnia audio processor.
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Separate multiband AGC and multiband limiter stages.
Most other audio processors take a shortcut in the execution of multiband AGC and limiting functions;
they put both functions within the same audio processing “block”. Omnia takes a more comprehensive
approach. We designed the multiband AGC and multiband limiting blocks completely separate from
each other. This allows us to give slightly different treatment as needed for the absolute cleanest and
competitively loudest audio. For example, some bands of limiting are best served by feed-back servo
control, while the higher bands are best served by feed-forward servo control. Our design, which breaks
these multiband functions into individual processing blocks, allows for the absolute best treatment of
each audio band.
Omnia ONE’s Powerful and Comprehensive Clipper
In any FM audio processor, the final audio clipper presents the largest hindrance or benefit to the
loudness and clarity of the on-air sound for a given modulation level. The FM clipper section in the Omnia
ONE, originally available only in big brother Omnia-6, includes two distinct sections: a bass-management
clipper and a main clipper. The bass clipper is quite sophisticated in its own right, but the main clipper
offers two different clipping styles and the ability to “balance” between them, if desired. This flexibility
gives the curious or competitive user the ability to finely tune this most important function.
Sensus Overview
Until now, digital signal processing has been a more precise numeric implementation of well-known
analog methods. Even relatively recently designed digital audio processors couldn’t veer too far from the
comparatively simplistic concepts that analog dynamics processing had utilized...until now!
Extremely high power DSP chips have become available and at relatively low cost, and they make it
possible to build smarter and more complex processing algorithms that were too difficult or impossible
(or too expensive) to do in the past.
Running on a platform of the latest high power DSP chips, the Omnia ONE and our new Sensus®
technology takes digital dynamics processing into a completely new frontier. Instead of the twodimensional static processing architecture of the past, Sensus® enables the audio processor to modify
its own architecture in real time and in response to ever-changing program content. Simply stated,
Sensus® has the ability to “sense” what must be done to a signal in order to best tailor it for output
to a codec. As program content changes, it “rearranges the algorithms” to accomplish this goal. The
uniqueness of the Sensus® technology makes it highly suitable not only for codec pre-conditioning (or
provisioning), but also for a range of other highly specialized signal processing challenges. The following
is a discussion of how Sensus® technology can be applied to a coded audio environment.
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Codec Provisioning
The codec is now a common denominator in the world of audio and broadcasting. Digital broadcasting
(HDTV, HDRadio, DAB, DRM), podcasting, webcasting, cellcasting, and downloadable music files all
employ a form of codec-based data compression in order to minimize the bandwidth required to
transmit audio data. The necessarily low bitrates utilized by these mediums presents a tough challenge
for any audio processor used prior to a codec. Traditional dynamics processors are designed to fulfill the
requirements of a medium where the functions are generally static. That is, they’re well suited to the
rather simplistic peak control and bandwidth limiting methods required for analog broadcasting, as well
as for the signal normalization techniques used in recording and mastering. Audio codecs on the other
hand are moving targets - each codec algorithm has its own set of artifacts.
Not only does the sonic quality vary depending on the algorithm and bitrate used, but more importantly
they vary in their ability to mask their own coding action. This is why we call it a ‘moving target’ , and
is why conventional audio processors fall short in a coded audio environment and can actually make
coding artifacts worse due to their inability to adapt appropriately to the changing operation of the
codec as the program content changes. Prior art in audio dynamics processing could only address some
of the challenges of provisioning audio for coding. This hurdle existed because the codec adapts to the
incoming program (so as to generate the least amount of output data representing the input audio)
causing the sonic artifacts generated by the process to continually change. Unless the audio processor
can predict these changing characteristics of the codec, it can’t possibly create output audio that is
perfectly tailored for the coding process.
Conventional processors utilize rather simplistic high frequency limiters and fixed low pass filtering that
does not change with the program material. When these less intelligent processors feed a codec the
audio might sound acceptable one moment and offensive the next. Because they cannot “know” what
the codec will do next, the result is over-compensated, dull and lifeless audio... audio that still contains
objectionable codec-generated artifacts!
Omnia ONE Multicast and HD Radio®
The advent of HD Radio has introduced the capability to transmit multiple program streams, or
“Multicast”, within a single 96kbps digital broadcast data channel. To facilitate this, multicast relies on
the use of codecs with comparatively low bitrates. A broadcaster can choose to transmit a number of
multicast channels and select the bitrate for each one. However, the more multicast channels there
are, the lower the bitrate each channel must have in order for them to all fit within the total available
bandwidth. To achieve maximum sound quality, the kind that attracts and holds listeners, those
channels need specialized dynamics processing capable of creating great sound regardless of program
content and bitrate. They need Sensus®.
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SPECIFICATIONS
Omnia ONE FM & SG
Frequency Response
• Complies with the standard 50 or 75 microsecond pre-emphasis curve within ± 0.50 dB, 30 Hz to 15
kHz. The analog left/right outputs and AES/EBU Digital outputs can be configured for flat or preemphasized output.
System Distortion
• Less than 0.01% THD, 20 Hz – 7.5 kHz. Second harmonic distortion above 7.5 kHz is not audible in the
FM system.
• *Signal-Noise Ratio: > -80 dB de-emphasized, 20 Hz –- 15 kHz bandwidth, referenced to 100%
modulation).
*The measured noise floor will depend upon the settings of the Input and Output Gain controls and
is primarily governed by dynamic range of the Crystal Semiconductor CS5361 A/D Converter which is
specified as >110 dB. The dynamic range of the internal digital signal processing chain is >144 dB.
Stereo Separation
• Greater than 65 dB, 20 Hz –- 15 kHz; 70 dB typical.
Crosstalk
• > -70 dB, 20 Hz -- 15 kHz.
Composite Outputs
• Source Impedance: 5 ohms or 75 ohms, jumper-selectable. Single-ended and floating over chassis
ground.
• Output Level: 0V to 10V in 0.05V steps, software adjustable.
• D/A Conversion: Texas Instruments/Burr Brown PCM1798, 24-bit sigmadelta converter.
• Configuration: Two electrically independent outputs. Software based level adjustment.
• Load Impedance: 50 ohms or greater load is suggested.
• Pilot Level: Adjustable from 4.0% to 12.0% in 0.1% steps and OFF.
• Pilot Stability: 19 kHz, ± 0.5 Hz.
• Signal-to-Noise Ratio: -85 dB typical, 75 µS de-emphasized, 15 kHz bandwidth, referenced to 100%
modulation).
• Distortion: < 0.02% THD 20 Hz – 15 kHz bandwidth, 75 µS deemphasized, referenced to 100%
modulation.
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• Stereo Separation: > 65 dB, 30 Hz – 15 kHz.
• Linear Crosstalk: > -80 dB, main to sub or sub to main channel (referenced to 100% modulation).
• Non-linear Crosstalk: > -80 dB, main to sub or sub to main channel (referenced to 100% modulation).
• 8 kHz Suppression: > 70 dB (referenced to 100% modulation).
• 76 kHz Suppression: > 80 dB (referenced to 100% modulation).
• Pilot Protection: > -65 dB relative to 9% pilot injection, ± 1 kHz. 57 kHz (RDS/RBDS) Protection: better
than -50 dB.
• Connectors: Two EMI suppressed female BNC, floating over chassis ground.
• Maximum Load Capacitance: 5nF (at 10 ohms source impedance).
• Maximum cable length: 100 feet/30 meters RG-58A/U.
Analog Audio Input
• Left/Right Stereo.
• Electronically balanced.
• Input impedance 10k ohms resistive.
• Maximum Input Level: +22 dBu.
• Nominal Input Level: +4dBu, which nets a -18dBFS input meter reading on a steady-state signal when
the Input Gain control is set to 0.0dB. Program material with a nominal average level (VU reading) of
+4dBu will typically produce peak readings on the input meter in the range of -12 dBFS to -6dBFS.
This is the correct operating level.
• A/D Conversion: Crystal Semiconductor CS5361, 24 bit 128x oversampled delta sigma converter with
linear-phase anti-aliasing filter.
• Pre-ADC anti-alias filter, with high-pass filter at <10 Hz.
• Connectors: Two, EMI-suppressed XLR-female. Pin 1 chassis ground, Pin 2 “Hot”.
Analog Audio Output
• Left/Right Stereo. Electronically balanced.
• Output Impedance 20 ohms.
• Minimum load Impedance: 600 ohms.
• Output Level adjustable from -2 dBu to +22dBu peak in 0.1dB steps.
D/A Conversion
• Crystal Semiconductor CS4391, 24 bit, 128x oversampled.
• Connectors: Two, EMI-suppressed XLR-male. Pin 1 chassis ground, Pin 2 “Hot”
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Digital Audio Input
• Configuration: Stereo per AES/EBU standard, CS8420 Digital Audio Transceiver with 24 bit resolution,
software selection of stereo, mono from left, mono from right or mono from sum.
• Automatically accepts and locks to input sample rates between 32 and 108 kHz.
• Connector: EMI-suppressed RJ-45 female pinned according to StudioHub+® standards. Transformer
isolated, balanced, and floating according to AES3 standard.
Digital Audio Output
• Stereo per AES3 standard. Digital Output sample rate can lock to the input, lock to an additional
external sync source, or use the internal 48kHz rate.
• Connector: EMI-suppressed RJ-45 female according to Studio- Hub+® standards. Transformer
isolated, balanced, and floating according to AES3 standard.
Digital Output Level
• -24.0 to 0.0 dBFS peak, software adjustable in 0.1dB steps.
External Sync Input
• External Sync: Allows the output sample rate to be synchronized to an AES3 signal applied to the Ext.
Sync input connector. (Does not accept Word Clock inputs)
• Connector: EMI-suppressed RJ-45 female according to Studio- Hub+® standards. Transformer
isolated, balanced, and floating according to AES3 standard.
External Sync Range
• Automatically accepts sample rates between 32kHz and 96kHz.
• Connector: EMI-suppressed RJ-45 female pinned according to StudioHub+® standards, Transformer
isolated, balanced, and floating according to AES3 standard.
Remote Control Methods
• Ethernet: TCP/IP control via web page interface and Java (TM) remote control program included in the
web pages. All software is served from the built-in web server; there is nothing to install on the user’s
computer.
• Ports Used: The defaults are TCP Ports 4545 and 4546 (for control and metering data, respectively
• Connectors: Modem port - EMI-suppressed DB-9 male connector. Ethernet - Industry standard EMIsuppressed RJ-45 connector.
GPI Interface
• Connector: EMI suppressed DB-9 female connector.
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Power Requirements
• Voltage: 100-250 VAC, 47-63 Hz., Less than 40 VA.
• Power Connector: EMI suppressed IEC male. Detachable 3-wire power cords supplied for US and
European use.
• Power Supply: Internal. Overvoltage and short circuit protected. Meets EN55022, EN55011 Level
B Conducted Emissions. EN61000-4-2, -3, -4, -5, -6 level 3 immunity compliant. Full international
safety approval. CE marked.
• Environmental: Operating Temperature: 32 to 122 deg. F / 0 to 50 deg. C for all operating voltage
ranges. Humidity: 0-95% RH, non-condensing.
Omnia ONE AM
Notes
• Discrete I/O measurements have been made in “Bypass” mode (available in the Input/Output menu).
• All measurements made with the supplied “FACT_TEST” preset, which is available in the Preset
Submenu.
System Frequency Response
• Complies with the NRSC emphasis curve within ± 0.50 dB, 30 Hz to 10 kHz. (At a setting of “10” on the
HF EQ control)
System *Signal to Noise Ratio
-80 dB de-emphasized, 20 Hz – -10 kHz NRSC bandwidth, referenced to 100% modulation).
*The measured noise floor will depend upon the settings of the Input and Output Gain controls and
is primarily governed by dynamic range of the Crystal Semiconductor CS5361 A/D Converter which is
specified as >110 dB. The dynamic range of the internal digital signal processing chain is >144 dB.
System Distortion
• Less than 0.01% THD, 20 Hz – 5 kHz. (second order harmonic
• distortion above 5 kHz is not relevant in the AM system due to the removal of harmonics by the
system’s 10 kHz low pass filter)
System Stereo Separation
• Greater than 65 dB, 20 Hz –- 10 kHz; greater than 70 dB typical.
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Analog Audio Input
• Left/Right Stereo.
• Electronically balanced.
• Input impedance 10k ohms resistive.
• Maximum Input Level: +22 dBu.
• Nominal Input Level: +4dBu, which nets a -18dBFS input meter reading on a steady-state signal when
the Input Gain control is set to 0.0dB. Program material with a nominal average level (VU reading) of
+4dBu will typically produce peak readings on the input meter in the range of -12 dBFS to -6dBFS.
This is the correct operating level.
A/D Conversion
• Crystal Semiconductor CS5361, 24 bit 128x over-sampled delta sigma converter with linear-phase
anti-aliasing filter. Pre-ADC anti-alias filter, with high-pass filter at <10 Hz.
• Connectors: Two, EMI-suppressed XLR-female. Pin 1 chassis ground, Pin 2 “Hot”.
Analog Audio Output
• Left/Right Stereo. Electronically balanced.
• Output Impedance 20 ohms.
• Minimum load Impedance: 600 ohms.
• Output Level adjustable from -2 dBu to +22dBu peak in 0.1dB steps.
• D/A Conversion: Crystal Semiconductor CS4391, 24 bit, 128x oversampled.
• Connectors: Two, EMI-suppressed XLR-male. Pin 1 chassis ground, Pin 2 “Hot”.
Digital Audio Input
• Configuration: Stereo per AES/EBU standard, CS8420 Digital Audio Transceiver with 24 bit resolution,
software selection of stereo, mono from left, mono from right or mono from sum.
• Automatically accepts and locks to input sample rates between 32 and 108 kHz.
• Connector: EMI-suppressed RJ-45 female pinned according to StudioHub+® standards. Transformer
isolated, balanced, and floating according to AES3 standard.
Digital Audio Output
• Stereo per AES3 standard. Digital Output sample rate can lock to the input, lock to an additional
external sync source, or use the internal 48kHz rate.
• Connector: EMI-suppressed RJ-45 female according to Studio- Hub+® standards. Transformer
isolated, balanced, and floating according to AES3 standard.
Digital Output Level
• -24.0 to 0.0 dBFS peak, software adjustable in 0.1dB steps.
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External Sync Input
• External Sync: Allows the output sample rate to be synchronized to an AES3 signal applied to the Ext.
Sync input connector. (Does not accept Word Clock inputs)
• Connector: EMI-suppressed RJ-45 female according to StudioHub+® standards. Transformer isolated,
balanced, and floating according to AES3 standard.
External Sync Range
• Automatically accepts sample rates between 32kHz and 96kHz.
• Connector: EMI-suppressed RJ-45 female pinned according to StudioHub+ standards, Transformer
isolated, balanced, and floating according to AES3 standard.
Remote Control Methods
• Ethernet: TCP/IP control via web page interface and JavaTM remote control program included in the
web pages. All software is served from the built-in web server; there is nothing to install on the user’s
computer.
• Ports Used: The defaults are TCP Ports 4545 and 4546 (for control and metering data, respectively)
• Connectors: Serial port - EMI-suppressed DB-9 male connector.
• Ethernet - Industry standard EMI-suppressed RJ-45 connector.
GPI Interface
• Connector: EMI suppressed DB-9 female connector.
Power Requirements
• Voltage: 100-250 VAC, 47-63 Hz., Less than 40 VA.
• Power Connector: EMI suppressed IEC male.
• Detachable 3-wire power cords supplied for US and European use.
• Power Supply: Internal. Overvoltage and short circuit protected. Meets EN55022, EN55011 Level
B Conducted Emissions. EN61000-4-2, -3, -4, -5, -6 level 3 immunity compliant. Full international
safety approval. CE marked.
Environmental
• Operating Temperature: 32 to 122 deg. F / 0 to 50 deg. C for all operating voltage ranges. Humidity:
0-95% RH, non-condensing.
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OMNIA | OMNIA ONE
Omnia ONE Multicast/DAB & Studio Pro
Note
• All measurements made using “Bypass” mode, which is available in the Input/Output menu.
General Audio Specifications
• Frequency Response: ± 0.50 dB, 20 Hz to 20 kHz with high pass filter disabled.
• Distortion: Less than 0.05% THD 20 Hz – 20 kHz bandwidth.
• *Signal-Noise Ratio: Greater than -100 dB, 20 Hz –- 20 kHz bandwidth, referenced to 0dBfs
*The measured noise floor will depend upon the settings of the Input and Output Gain controls and is
primarily governed by dynamic range of the Crystal Semiconductor A/D Converter which is specified as
>100 dB. The dynamic range of the internal digital signal processing chain is >144 dB.
Stereo Separation
• Greater than 80 dB, 20 Hz –- 20 kHz; 90 dB typical.
Analog Audio Input
• Left/Right Stereo.
• Electronically balanced.
• Input impedance 10k ohms resistive.
• Maximum Input Level +24 dBu.
• Nominal Input Level: +4dBu (A +12dBu input results in –12dBFS input meter reading with Input Gain
set to 0.0dB. A 0dBu input signal results in a -12dBFS input level when Input Gain is +12dB.)
A/D Conversion
• Crystal Semiconductor 24 bit 128x oversampled delta sigma converter with linear-phase anti-aliasing
filter. Pre-ADC anti-alias filter, with highpass filter at <10 Hz.
• Connectors: Two, EMI-suppressed XLR-female. Pin 1 chassis ground, Pin 2 “Hot”.
Analog Audio Output
• Left/Right Stereo. Electronically balanced.
• Output Impedance 20 ohms.
• Minimum load Impedance 600 ohms.
• Output Level adjustable from -2 dBu to +22dBu peak in 0.1dB steps.
• D/A Conversion: Crystal Semiconductor CS4390 24 bit, 128x oversampled.
• Connectors: Two, EMI-suppressed XLR-male. Pin 1 chassis ground, Pin 2 “Hot”.
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OMNIA | OMNIA ONE
Digital Audio Input
• Configuration: Two-channel stereo per AES3 standard via CS8420 Digital Audio Transceiver with
24-bit resolution. Software selection of stereo, mono from left, mono from right or mono from sum.
Automatically accepts sample rates between 24 kHz and 96 kHz.
• Connector: EMI-suppressed RJ-45 female pinned according to StudioHub+® standards. Transformer
isolated, balanced, and floating according to AES3 standard.
Digital Audio Output
• Stereo per AES3 standard. Digital Output sample rate software selectable for internal 48kHz,
synchronize to AES input, or synchronize to auxiliary AES sync input (per AES-11 / DARS).
• Connector: EMI-suppressed RJ-45 female according to Studio- Hub+® standards. Transformer
isolated, balanced, and floating according to AES3 standard.
Digital Output Level
• -22.0 to 0.0 dBFS peak, software adjustable in 0.1dB steps.
Digital Sync Input
• Output sample rate can be synchronized to the signal present on the AES/EBU input or to the AES3
signal applied to the Ext. Sync connector.
External Sync Range
• Accepts 32kHz to 96 kHz for synchronization of the Digital
• Output signal to an external reference. Automatically accepts sample rates between 32kHz and
96kHz.
• Connector: EMI-suppressed RJ-45 female pinned according to StudioHub+® standards, Transformer
isolated, balanced, and floating according to AES3 standard.
Remote Control Methods
• Ethernet: TCP/IP control via web page interface and JavaTM remote control program included in the
web pages. All software is served from the built-in web server; there is nothing to install on the user’s
computer.
• Ports Used: The defaults are TCP Ports 4545 and 4546 (for control and metering data, respectively).
• Connectors: Serial port - EMI-suppressed DB-9 male connector.
• Ethernet - Industry standard EMI-suppressed RJ-45 connector.
GPI Interface
• Connector: EMI suppressed DB-9 female connector.
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OMNIA | OMNIA ONE
Power Requirements
• Voltage: 100-250 VAC, 47-63 Hz. Less than 25 VA.
Power Connector
• EMI suppressed IEC male.
• Detachable 3-wire power cords supplied for US and European use.
Power Supply
• Internal. Overvoltage and short circuit protected. Meets EN55022, EN55011 Level B Conducted
Emissions. EN61000-4-2, -3, -4, -5, -6 level 3 immunity compliant. Full international safety approval.
CE marked.
Environmental
• Operating Temperature: 32 to 122 deg. F / 0 to 50 deg. C for all operating voltage ranges. Humidity:
0-95% RH, non-condensing.
Regulatory
North America: FCC and CE tested and compliant, power supply is UL approved.
Europe: Complies with the European Union Directive 2002/95/EC on the restriction of the use of certain
hazardous substances in electrical and electronic equipment (RoHS), as amended by Commission
Decisions 2005/618/EC, 2005/717/ EC, 2005/747/EC (RoHS Directive), and WEEE.
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OMNIA | VOCO 8
Omnia VOCO® 8
Up to eight individually processed mics
networkable through an entire facility
OVERVIEW
VOCO 8 is more than just a simple mic processor--It is a complete solution for managing microphone
audio throughout your entire facility with a comprehensive set of tools such as:
• “Dominate-It” where the host mic can always be the dominant voice.
• Multiband processing
• Studio grade mic preamps with phantom power
• Eight line-level inputs
• “Session Recall” for convenience
• Livewire+™ AES67 support
The Omnia VOCO can adapt to nearly any voice or microphone, with factory presets tuned for both male
and female voices and various degrees of processing.
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FEATURES
Processing
• De-Esser
• 3-band Noise gate
• 3-band Processing
• 4-Band EQ
• Brick Wall Limiter
Processing Chain Extra Features
• Low Pass / High filters
• Phase rotation
• Dominate-It (reduces the level of other microphones by a predefined level whenever microphones
with the Dominate-It feature are active.)
• Dual mix buses
• Preset sharing between networked VOCO units
• Share one VOCO between multiple studios
• Session Recall
• Control via external automation systems with “Link & Share”
• 192 kHz Sampling Rate for Processing
• Processing latency as low as 3 ms
Inputs
Eight high quality microphone preamplifiers, each switchable to line level analog input, four stereo AES/
EBU inputs (eight mono), Livewire+™ AES67 AoIP input
Outputs
Line level analog, AES/EBU, and Livewire+™ AES67 AoIP
Bus Mix
Mix multiple microphones onto one of two mix busses to simplify console channel allocation, or use
channels independently.
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OMNIA | VOCO 8
IN DEPTH
A User Friendly Control Interface
• Control all Mics on one screen
• Change all settings from one single screen. No more opening and closing windows to go from one
function to another.
• “Basic” mode and “Advanced” mode: “Basic”, allows selection of sessions and presets without direct
access to processing parameters. “Advanced” mode, gives full access to all processing parameters
in a single interface. Input and output VU meters along with de-esser and gate LEDs give access to
channel activity at a glance.
• Unlimited “Undo/Redo” tracks all changes and allows them to be saved as new presets.
• “Compare” function allows quick comparison against a “reference” preset.
• Omnia Remote Gateway software works with Microsoft Windows XP SP3, Windows 7 (32 and 64 bit),
Server 2008 R2, and Debian Linux.
GUI #1: Studio Mode GUI
“Studio Mode” gives access to all processing parameters for all 8 channels in a unified interface for
“power users” who want the ultimate in control.
GUI #2: Live Mode GUI
“Live Mode” allows quick selection of sessions and presets in just a few clicks for fast-paced live show
environments without giving direct access to processing parameters.
HQSound 192 kHz
Omnia VOCO 8 features the HQSound 192 kHz algorithm. This high resolution algorithm provides a large
range of gain control without the typical “pumping” or “smashed” sound that can occur with high levels
of compression. The result is a robust microphone sound sure to please even the most critical ears.
Effective 3-band noise gate
In voice processing, to get an efficient noise gate on all voices with one preset is impossible. This is
mainly due to differences in levels and consistency between voices. With VOCO 8 it is possible to create
a preset for each talent. This is a key point for a perfect noise gate efficiency. Working in 3-band is a real
advantage. In noise gate, bands are able to work independently or in a Master/Slave scenario. This helps
to isolate noise coming from table and doors.
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S.I.S – Sound Impact System
A part of the HQSound 192 algorithm, S.I.S preserves attacks automatically for maximum voice impact.
Preset Sharing
Another unique feature, preset sharing allows users to synchronize presets and all changes on an
unlimited group of VOCO 8 units. Preset sharing will automatically update all VOCO 8 processors without
the need to load presets on each unit manually. In addition, whenever a new preset for a specific host is
created in one studio, it will automatically be available in all other studios.
Multi-Studio Management
The Omnia VOCO 8 can process up to 8 microphones. Thanks to Multi-Studio mode, the Omnia VOCO 8 can
distribute these resources across several studios. For example, if you have two studios each with three
microphones, a single VOCO 8 can be used to independently handle the microphones for each studio.
Each studio can also save and recall its own sessions for rapid configuration.
Security
The VOCO 8 features a full set of security features to prevent unauthorized tampering with presets and
configuration. Multiple users and access levels are supported across all studios.
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SPECIFICATIONS
Mic Input
• 8 channels, XLR
• +48v phantom, switchable
• Source impedance: 150 Ohms
• Input impedance: 4000 Ohms
• Level Range: -75 dBu to -20 dBu
Line Level Input
• ¼” (6.33mm)
• Level: +4dBu or -10dBu
Digital Input
• Quantity: 4 stereo (2 channels per AES/EBU input)
• Standard: AES/EBU
• Sampling Rate: 32 to 192 kHz - 24 bits
• DB-25 using Tascam® format
Livewire+™ AES67 Input
• Quantity: 8
• Type: Livewire (Standard or Live stream) & AES67
• Level: Adjustable in Omnia VOCO user Software
• Connector: Ethernet 100BASE-T
AES/EBU Sync
• Internal digital sync from high precision clock source.
• External reference supported from any AES/EBU input port.
Digital Output
• Quantity: 4 stereo (2 channels per AES/EBU output)
• Standard: AES/EBU
• DB-25 using Tascam format
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OMNIA | VOCO 8
Livewire+™ AES67 Output
• Quantity: 8
• Type: Livewire (Standard or Live stream) & AES67
• Level: Adjustable in Omnia VOCO user Software
• Connector: Ethernet 100BASE-T
GPI Interface
• Connector: Standard DB-15
Audio Performance
• Processing delay: 3 ms
• Frequency response: 10Hz - 22 kHz +/-0.2dB
• Distortion: <0.2% THD
Compatible Operating System for Remote Control Software
• Omnia Remote Gateway software works with Microsoft Windows XP SP3, Windows 7 (32 and 64 bit),
Server 2008 R2, and Debian Linux.
Omnia VOCO to Client Communication Interface
• TCP/IP: Client (Remote via Ethernet)
• Link & Share: 100% of parameter are accessible through telnet protocol
Power Requirements
• 100-264 VAC, 47-63Hz autosensing, dual PSU
Power Connector
• Dual IEC male, detachable 3-wire power cords supplied
Regulatory
North America: FCC and CE tested and compliant, power supply is UL approved.
Europe: Complies with the European Union Directive 2002/95/EC on the restriction of the use of certain
hazardous substances in electrical and electronic equipment (RoHS), as amended by Commission
Decisions 2005/618/EC, 2005/717/ EC, 2005/747/EC (RoHS Directive), and WEEE.Ethernet - Industry
standard EMI-suppressed RJ-45 connector.
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AXIA | FUSION
Axia Fusion®
Where Design and Technology Intersect.
OVERVIEW
Since 2003, Axia® has become the name broadcasters think of first when they think of networked
broadcast facilities. Thousands of radio and audio professionals have made Axia their first choice for
powerful, flexible, easy-to-use mixing consoles.
Fusion is the new Axia modular console packed with features and capabilities refined from over a
decade’s worth of IP-Audio experience. It’s available in frame sizes to support consoles of 8 to 40 faders
in single or multiple linked frames. Fusion may be powered by the Axia PowerStation® or StudioEngine
DSP mixing engines, and connects to the Axia network with a single CAT-6 Ethernet cable, allowing the
sharing of local audio devices (and their associated GPIO control) among multiple studios to maximize
efficiency and reduce cost.
Fusion has four stereo Program buses, four Send buses, and two Return buses. A variety of module
types are available, from fader-only modules to Call Controller modules with integrated multi-line
controls for Telos® multi-line phone systems. Fusion also features unique Axia VMix (Virtual Mixer)
channels, which allow combining up to 5 audio sources for presentation on a single console fader —
further extending the flexibility and usefulness of the console.
Other features include Auto-assigned, auto-generated mix-minus on each channel, easy individual or
group talkback for remote talent cueing, one-button off-air phone record mode, and up to 99 Show
Profiles console “snapshots” for set, save and recall of console layouts customized to the working style
of individual shows or operators. Built-in digital EQ may be applied individually to all audio sources, as
well as dynamic microphone processing from Omnia® for all mic sources.
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AXIA | FUSION
Fusion Screen Shots:
The main view
AUX SEND view with BBC PPM ballistics shown on meters
Channel EQ view with VU meter ballistics shown
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Channel Mic processing view with VU meter ballistics shown
Loudness metering, EBU Digital ballistics shown on main meters, and 4 assignable meters
Show profile selection view with Nordic ballistics shown
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AXIA | FUSION
Channel source profile selection view with EBU Digital ballistics shown
FEATURES
• From 8 to 40 fader channels, each with instant, unlimited access to any source. Assign any type of
source to any channel.
• Rugged construction of extruded aluminum ensures rigidity and EM-tightness. Anodized aluminum work
surface with laser-etched markings that can’t be rubbed off ensures durability and good looks for life.
• Four main stereo outputs (Program-1 through Program- 4), plus four stereo Aux sends and
two Aux returns.
• High-resolution OLED displays above each fader strip display selected source, full-time source and
backfeed confidence metering, talkback status, pan/fade information and more.
• Integrated intercom capability includes built-in IFB for two-way communication to individual talent
positions via headphone feeds and mics, plus a variety of optional drop-in intercom modules that
connect to Axia IP Intercom whole-plant intercom systems.
• Flexible, intuitive Talkback system lets board ops talk to hosts, studio guests, external feeds — any
source with an associated backfeed.
• Every channel has a stereo Preview (“cue”) function, with a unique latching interlock system for fast,
intuitive operation. Multiple channels may be assigned to Preview simultaneously.
• Reconfigurable monitor section with reassignable controls let operators instantly change monitored
sources “on-the-fly.”
• Software control of options such as EQ, mic dynamics, aux sends and returns, pan and balance and
other features delivers maximum flexibility without panel clutter or intimidating controls.
• Built-in Omnia dynamics processing lets operators combine compression, de-essing and expansion
with EQ to “sweeten” microphone sources.
• A unique, flexible Record Mode enables one-button setup of record mixes for phone bits or off-air
interviews.
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AXIA | FUSION
• Consolidated user display conveys meter, clock, timer and monitor source information at a single
glance. Meter up to 6 buses at once by default, using VU or PPM-style ballistics — add another 4
meters for a total of 10 if desired.
• Precision timer and clock functions, including an event timer that can be triggered by pre-defined
sources, a countdown timer with last-minute alerting and a time-of-day clock that can be
synchronized to network time using NTP.
• Show Profiles set-save-recall feature allows users to instantly recall a customized personal profile,
or a profile tailored to specific show types. Up to 99 Show Profiles can be saved for interview shows,
music-intensive programming, call-in talk shows, etc.
• Console functions can be accessed remotely for configuration, management and diagnostic purposes
using any standard Web browser.
• Built-in 5.1 discrete mixing capabilities for production use.
• Optional Telos phone control module provides direct, on-the-console line switching control of any
Telos multi-line broadcast phone system.
• Numeric keypad (with # and * keys) lets operators quickly place calls with phone systems or codecs
attached to the Axia network.
• Completely automatic, hands-free mix-minus generation for every Phone caller or remote Codec source.
• Built-in User Keys for can be programmed with Axia PathfinderPC routing control software to control
profanity delay units (such as the 25-Seven® Program Delay Manager), or can be used to trigger
routing salvos, scene changes or send GPIO commands.
• No audio passes directly through the Fusion control surface, keeping your programming safe from
studio accidents. All mixing and processing is performed by the StudioEngine or PowerStation
mixing engines.
• Axia’s trademark long-life conductive-plastic faders with side-loading actuation defy dirt, grit and dust.
• Aircraft-grade switches with LED lighting have been tested to withstand millions of operations.
• Can be directly remote-controlled using Axia SoftSurface software for Windows.
• Fusion 3.1 software update adds AES67 support.
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IN DEPTH
A decade of IP-Audio experience, packed into a single console.
Ask broadcasters who’s the leader in IP-Audio, and chances are good they’ll tell you “it’s Axia.” That’s
because we’re the originators of studio networking for broadcast facilities; we produced our first
console in 2002 (back when everyone else was saying “Audio over Ethernet? That’ll never work!”), and
we’ve been listening, learning and inventing ever since. That’s why thousands of happy broadcasters
worldwide have put Axia consoles to work for them.
Now, we’d like to introduce you to Fusion, the newest modular IP-Audio broadcast console from Axia.
Why did we name it Fusion? According to Webster’s, “fusion” means “joining two or more things
together to form a single entity.” And that’s just what we’ve done. We’ve taken everything we’ve
learned in the past decade – about talent experiences, on-air mechanics, in-studio workflow and
more – and combined those thoughts, ideas and observations into the smoothest, most intuitive, most
indestructible networked console yet.
Fusion was created from what you, our clients, have taught us about today’s fast-paced broadcast
environment; an environment that interfaces with listeners both on the dial and on the ‘Net. And our
talented team (made up of scientists, engineers and even former air talent) built a console that has what
it takes to support everything from the mile-a-minute call-in talk shows, to tight music-driven formats,
to multi-talent morning shows — or anything else your programming department can imagine.
So, what things did we learn that made their way into Fusion? That “Powerful” doesn’t have to mean
“complicated,” for one. You taught us that a broadcast console can have lots of capabilities, and still be
easy to use. So our console designers looked at the way broadcasters accomplish complicated things,
and figured out ways to make them simpler.
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Like mix-minus: Fusion creates mix-minus for every codec and phone caller you put on the air.
Automatically. With no extra buttons to push, bus assignments to make, or settings to change.
It just happens.
Or, take recording off-air phone bits for later playback. ‘Til now, it meant taking talent and callers from
Program buses manually, assigning them to utility buses, manually starting a recorder…and don’t
forget creating the mix-minus! Fusion’s one-touch Record Mode does all of that for you, managing bus
assignments, creating the mix-minus, recording the conversation, even changing the monitor feed —
then putting it all back the way it was when you’ve finished.
Then there’s Fusion’s built-in talent Mic processing that combines with its Show Profile set-saverecall system. Got a jock whose voice needs a little extra sweetening? Build a custom voice-processing
mode for them using Omnia compression, limiting and de-essing tools, then save those settings in a
personalized profile they can recall anytime they want.
What about reconfiguring the console from producing a music show to handling a live in-studio band
performance with multiple mics and DI inputs? In the old days, you’d be pushing dozens of buttons to
make new input choices, bus assignments, monitor settings and EQ tweaks. With Fusion, you can do it
all with two clicks, bringing up a stored Show Profile snapshot that suits the job at hand.
In fact, Fusion is a power-user’s dream. A unique Expert Source Profile mode lets you build custom
audio inputs with completely customized GPIO functions, IFB backfeed and mix-minus, and Monitor and
Program Bus assignments — all based on channel On/Off/Preview state. These powerful tools give
you complete control of the behavior of audio sources – on a per-show basis – as they enter and leave
the console, allowing automation of complex operations and helping operators run easier, more errorfree shows.
Mixing capacity? Fusion has 4 Program buses, plus 4 Aux sends and 2 Aux returns, along with 16 fivechannel “Virtual Mixers” that let you mix multiple audio inputs using software-controlled virtual faders.
You also told us you value durability. Axia consoles are known for their toughness, and Fusion is no
exception. All work surfaces are made of heavy-duty, anodized machined aluminum panels. This ensures
that Fusion will shrug off mistreatment by even the toughest jocks — there are no plastic overlays to
crack and peel, and no paint to wear off. Fusion’s markings are laser-etched, so they stay legible forever;
they literally can’t rub off!
We also put razor-sharp, high-resolution OLED displays on every fader strip. These are readable from
nearly every angle, which aids talent during fast-paced show production. These OLEDs also display
full-time confidence meters for each assigned source, further ensuring smooth, error-free shows and
helping prevent dead air. You’ll find even more information in Fusion’s on-screen metering display, too.
Making the most of modern wide-screen monitor design, Fusion displays six stereo meters by default,
making it perfect for main control rooms with multiple active program outputs. Meters are switchable
and can display gain in VU, DIN or BBC-style PPM, plus EBU Digital and Nordic scales.
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Finally, our designers have made Fusion the easiest Axia console, ever, to set up. Like any other AoIP
device, it connects directly to the network via Ethernet, allowing unparalleled flexibility when placing
consoles and connecting to mixing engines.
Take a good look at the bottom of a fader strip. Notice there’s a “Talkback” key there? Every mic source
on your Axia network – news booth, talent position, producer’s station – can have an individual
headphone backfeed. Touching the Talkback key associate with any mic source allows your board op to
talk privately to just that talent position (and they can talk back to the CR, too, using their mic). Fusion
operators can use even use this unique Talkback feature to communicate with phone callers, remote
talent or other studios using the console mic, and can “button mash” to communicate with entire groups
of locations at once.
With Fusion, controls for all of your studio devices are right on the console, where they’re most useful.
For instance: phone hybrid modules with dedicated faders give instant control of Telos talkshow
systems; talent can dial, answer, screen and drop calls without ever taking their eyes off the console,
which means smoother, easier on-air phone segments. IP Intercom modules let talent communicate
with other places in the broadcast plant using the CR mic, and even take broadcast-quality intercom
audio directly to air when desired.
Like all Axia consoles, Fusion has a built-in, password-protected Web server for easy remote
administration from your office, your boss’ office, even your home office – anywhere there’s a network
connection. Fusion works with Axia SoftSurface virtual console software too, so talent can take direct
remote control of the console.
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Naturally, our designers put their legendary attention to detail to work on Fusion. Redundant power
supply options with fanless convection cooling, hot-swappable modules, silky-smooth conductiveplastic faders with a side-loading design that foils dirt and other contaminants, razor-sharp OLED
options displays, optical rotary encoders and, of course, avionics-grade switches with LED lighting
(tested to withstand more than five million operations) are all part of Fusion’s premium design. You
can also add redundant power without additional IO with Axia Console Power Supply, which offers a
single-cable connection to PowerStation Main, providing backup power with automatic switching. (Autosensing power supply, 90VAC to 240VAC, 50 Hz to 60 Hz. 250 Watts, 2RU.)
The only thing not premium about Fusion is...its price. For all of this power, flexibility and ease of use,
Axia clients have told us they’d expect to pay much more. Luckily, you don’t have to.
Fusion is fully customizable, of course, with a full options list of module types designed to suit your
station’s unique way of making great radio. There are integrated controls for phones, codecs and studio
talkback, SmartSwitch modules with context-sensitive displays that enable one-touch router salvos,
even motorized faders for remote control or integration with your delivery system.
4-Fader Module
The 4-Fader module is where you start to build your Fusion. Use it for any
source: line, mic, hybrid, phone or codec source. Comes in standard and motorized-fader versions for use with automation systems or other moving-fader
applications.
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Monitor Modules
The other basic module you’ll need is a Monitor module. You can choose
between two types:
The Expert Monitor/Navigation module shown here has extended monitor,
headphone and preview controls, a numeric entry/dialpad that can be used
with Fusion phone modules, plus four programmable User Keys that can
trigger GPIO commands like profanity delay controls or recording devices,
or be used with Axia PathfinderPC software to issue routing salvos, initiate
scene changes, etc.
For studios where expert monitor controls are not needed, the Standard
Monitor/Navigation module is a space-saving design that incorporates two
faders in addition to the numeric entry/dial pad and basic Monitor/Headphone
controls.
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Call Controller Module
The Call Controller module has two faders plus integrated line switching
controls with Status Symbols, for on-console control of advanced Telos
broadcast phone systems. Available in standard and motorized-fader versions.
Switch Modules
Two available styles of programmable switch modules work with Axia PathfinderPC
routing control tools. They make it easy to put custom routing salvos or simple machine
logic right at talent’s fingertips.
Economical 10-Button Film-Cap switch modules are perfect for giving talent access
to often-used machine-control or GPIO-triggered routing commands. LED button
backlights can be individually changed to any of 8 colors.
Need more complex control of routing functions? 10-button SmartSwitch modules
feature dynamic, backlit LCD displays. Button functions, colors and even text can be
programmed to change in response to user input using Axia PathfinderPC software.
Construct custom routing salvos, cascading machine-logic command sets, or other
complex routing operations.
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IP Intercom Modules
Fusion consoles come equipped with a sophisticated Talkback system that allows
board ops to communicate directly with remote talent via individual Talkback
channels. But when larger facilities require even more powerful communication
capabilities, these 10 and 20-station intercom modules, part of the Axia IP
Intercom system, put broadcast intercom controls right in the console. Station
presets and GPIO functions for both types of modules are programmed using any
standard Web browser. Using these Intercom modules, Fusion operators can
instantly talk to any other studio, control room, operations center – even PCs
equipped with Axia SoftCom intercom software. And the audio is broadcast-quality,
so putting an Intercom source on-air is easy and sounds great.
10 and 20-station OLED Intercom modules feature high-resolution
programmable OLED displays that indicate assigned stations. The 10-Station
Filmcap intercom module has 10 LED-lit film-cap buttons for economical onconsole IP Intercom integration.
Fusion User Panels
Fusion Mic Control / Headphone Selector Panel
In-studio accessory panel mounts in tabletop or turret. Provides remote control of mic
on/off functions; dedicated Mute and Talkback buttons give talent full control of their
position. Volume/selection knob allows users to select their headphone monitor
source; display readout confirms their choice. Connects to Fusion Interface Board
CANbus using CAT-5 cable. 6”x 2”, requires 2” mounting depth.
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Fusion Headphone Selector Panel
In-studio accessory panel with volume/selection knob mounts in tabletop or turret to
allow users to select their headphone monitor source; display readout confirms their
choice. Two preset buttons let talent quickly recall frequently-listened-to sources.
Connects to Fusion Interface Board CANbus using CAT-5 cable. 6”x 2”, requires 2”
mounting depth.
Fusion Mic Control Panel
In-studio accessory panel for remote control of mic on/off functions. Mounts in
tabletop or turret; includes dedicated Mute and Talkback buttons. Requires one free
Axia GPIO port per panel. 6”x 2”, requires 2” mounting depth.
Fusion Producer’s Mic Control Panel
Special accessory panel for producers source profiles, which mounts in tabletop or
turret to provide remote control of mic on/off functions. Includes dedicated Mute key.
Two Talkback keys allow producers to talk to control room board op, studio guests, or
codecs. Requires one free Axia GPIO port per panel. 6”x 2”, requires 2” mounting depth.
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Fusion 4-Button LCD SmartSwitch Panel
Program these backlit LCD button panels using Axia PathfinderPC software to provide
producers or talent with remote access to often-used machine control or software
functions. In-button LCD display shows function readout. Dynamic programming
allows specific assigned functions to change for each Show Profile or in response to
user activation. Requires PathfinderPC (P/N 3001-0015). Easily connects to Fusion
Interface board CANbus using CAT-5 cable. Mounts in tabletop or turret. 6”x 2”,
requires 2” mounting depth.
Mixing Engines
Fusion consoles were designed to give you maximum flexibility and configuration options. So instead
of just one mixing engine, you’ve got your choice of two! Pair your Fusion control surface with a
powerful Linux-based StudioEngine and separate xNode audio interfaces – or choose the PowerStation
integrated console engine, an all-in-one powerhouse with audio I/O, DSP mixing engine and integrated
zero-configuration network switch.
StudioEngine
Pair your Fusion with Axia StudioEngine, an extremely powerful mixing and processing device based
on a blazingly-fast Intel processor. Each StudioEngine is fanless, has dual-redundant field-replaceable
modular power supplies, and has so much CPU power it can outperform the very largest digital or
router-based consoles. StudioEngine has multiple simultaneous inputs, outputs, mix-minus feeds,
monitor signals, EQ and voice processing; it’s the power behind state-of-the-art broadcast studios from
New York to Tokyo.
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PowerStation
PowerStation is an “integrated console engine”, an all-in-one devices that makes it very easy to install
Axia studios and Fusion consoles. Inside that ruggedly handsome case you’ll find a super-powered
DSP mixing engine, husky power supply sourced from telecom gear designed for harsh environments,
plenty of built-in digital, analog and mic I/O, plus EQ, voice processing — and even a custom, built-forbroadcast Ethernet switch with Gigabit connectivity.
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SPECIFICATIONS
Microphone Preamplifiers
• Source Impedance: 150 Ohms
• Input Impedance: 4 k Ohms minimum, balanced
• Nominal Level Range: Adjustable, -75 dBu to -20 dBu
• Input Headroom: >20 dB above nominal input
• Output Level: +4 dBu, nominal
Analog Line Inputs
• Input Impedance: >40 k Ohms, balanced
• Nominal Level Range: Selectable, +4 dBu or -10dBv
• Input Headroom: 20 dB above nominal input
Analog Line Outputs
• 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 Ohms, 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
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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
Equivalent Input Noise
• Microphone Preamp: -128 dBu, 150 ohm source, reference -50 dBu input level
Total Harmonic Distortion + Noise
• Mic Pre Input to Analog Line Output: <0.005%, 1 kHz, -38 dBu input, +18 dBu output
• 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 kH
• Microphone channel to channel isolation: 80 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
• Microphone Input CMRR: >55 dB, 20 Hz to 20 kHz
Audio Processing
Equalizer
• Frequency Bands: 20Hz to 320Hz, 125Hz to 2KHz, 1.25KHz to 20KHz.
• Cut/Boost range on each band: -25dB to +15dB.
• Q-factor: Automatic - bandwidth varies based on amount of cut or boost.
Compressor
• Threshold: -30dB to 0dB Ratio: 1:1 to 16:1
• Post-processor Trim Level: Adjustable from -20dB to +20dB
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Expander/Noise Gate
• Threshold: -50dB to 0dB Ratio: -30dB to 0dB
De-esser
• Threshold: -20dB to 0dB Ratio: 1:1 to 8:1
Axia Console Power Supply
• Add redundant power to PowerStation main without additional IO.
• Single-cable connection to PowerStation main provides backup power with automatic switching.
• Auto-sensing power supply, 90VAC to 240VAC, 50 Hz to 60 Hz.
• Power consumption: 250 Watts.
Power Supply AC Input, PowerStation Aux & Main
• Auto-sensing supply, 90VAC to 240VAC, 50 Hz to 60 Hz, IEC receptacle, internal fuse
• Power consumption: 500 Watts
Operating Temperatures
• -10 degrees C to +40 degrees C, <90% humidity, no condensation
Regulatory
North America: FCC and CE tested and compliant, power supply is UL approved.
Europe: Complies with the European Union Directive 2002/95/EC on the restriction of the use of certain
hazardous substances in electrical and electronic equipment (RoHS), as amended by Commission
Decisions 2005/618/EC, 2005/717/ EC, 2005/747/EC (RoHS Directive), and WEEE.
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Element®
World’s Most Popular IP-Audio Console
OVERVIEW
The Axia® Element is the world’s most popular IP-Audio mixing console, in use at thousands of
broadcast facilities every single day. Element is a modular console. Frames are available in sizes from 8
to 28 positions, with support for up to 40 faders in multiple linked frames. The Element control surface
works with the Axia PowerStation® and StudioEngine DSP mixing engines, and connects to the Axia
network with a single CAT-6 Ethernet cable. The networked nature of Element (and all Axia mixing
consoles) allows sharing of local audio resources and associated GPIO control across multiple studios.
Element features four stereo Program buses, four Send buses, two Return buses and a number of
VMix (Virtual Mixer) channels which allow combining up to 5 audio sources for presentation on a single
console fader. A variety of module types provide control of mic/line inputs, telephones and other devices.
Enhanced, integrated features for phones and codecs include auto-assigned, auto-generated mix-minus
on each channel, easy individual or group talkback for remote talent cueing, one-button off-air phone
record mode, and optional integrated Telco line switching. Show Profiles allow console “snapshots” with
different preferences, layouts and defaults to be loaded instantly, customizing the board to each show or
talent if desired.
Additionally, Element includes digital EQ which may be applied individually to all audio sources, dynamic
microphone processing from Omnia® for all mic sources, and many other advanced features.
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FEATURES
• Supports up to 40 faders, each with instant, unlimited access to any source. You can assign any type of
source to any channel.
• Four main stereo outputs (Program-1 through Program- 4), plus four stereo Aux sends and two Aux
returns.
• 10-character alpha-numeric displays above each fader channel always show the current selected
source.
• Each channel is equipped with a Status Symbol™ display which provides talkback, mix-minus, and
other source-related communication information.
• Every channel has a stereo Preview (“cue”) function, with a unique latching interlock system for fast,
intuitive operation. Multiple channels may be assigned to Preview simultaneously.
• Reconfigurable monitor section with reassignable controls let operators instantly change monitored
sources “on-the-fly.”
• Flexible, intuitive talkback system lets board ops talk to hosts, studio guests, external feeds — any
source with an associated backfeed.
• Software control of options such as EQ, mic dynamics, aux sends and returns, pan and balance and
other features delivers maximum flexibility without panel clutter or intimidating controls.
• Built-in Omnia dynamics processing lets operators combine compression, de-essing and expansion
with EQ to “sweeten” microphone sources.
• A unique Record Mode enables one-button setup of record mixes for phone bits or off-air interviews.
• Consolidated user display conveys meter, clock, timer and monitor source information at a single
glance. Use any external VGA monitor you choose, from a 12” LCD to a DLP wall projector!
• Precision timer and clock functions, including an event timer that can be triggered by pre-defined
sources, a countdown timer with last-minute alerting and a time-of-day clock that can be
synchronized to network time using NTP.
• Show Profiles set-save-recall feature allows users to instantly recall a customized personal profile,
or a profile tailored to specific show types. Up to 99 Show Profiles can be saved for interview shows,
music-intensive programming, call-in talk shows, etc.
• Console functions can be accessed remotely for configuration, management and diagnostic purposes
using any standard Web browser.
• Built-in 5.1 discrete mixing capabilities for production use.
• Optional Telos® phone control module provides direct, on-the-console line switching control of any
Telos multi-line broadcast phone system.
• Numeric keypad (with # and * keys) lets operators quickly place calls with phone systems or codecs
attached to the Axia network.
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• Completely automatic mix-minus generation for every Phone caller or remote Codec source.
• Built-in control keys for external profanity delay unit integrate via Livewire® with 25-Seven® Program
Delay Manager, or can be slaved to any other PDU using standard GPIO closures.
• No audio passes directly through Element — all mixing and processing is performed by the
StudioEngine or PowerStation mixing engines. Console connects to the Axia network using just one
cable.
• Long-life conductive-plastic faders with side-loading actuation defy dirt, grit and dust.
• Aircraft-grade switches with LED lighting have been tested to withstand millions of operations.
• Modules are available in choice of Bronze or Silver, with high-impact Lexan overlays. Custom-designed
fader and switch surrounds prevent cracking, chipping or peeling; markings can’t fade or rub off — ever.
• Can be directly remote-controlled using Axia SoftSurface software for Windows.
• Fusion 3.1 software update adds AES67 support.
IN DEPTH
The Choice of Connected Broadcasters Everywhere.
Axia was launched by Telos in 2003 to make digital mixing consoles. But we had a unique vision:
Axia consoles would be integrated with the routing switcher, and networked to share resources and
capabilities throughout the studio complex. Using this intelligent network of studio devices, talent
would benefit from consoles more powerful and easier to use than ever. 10 years and more than 6,000
studios later, broadcasters have made Axia consoles the most popular networked consoles in the world,
powering studios around the globe for the world’s most demanding broadcasters.
So, why have broadcasters made Element the world’s most popular IP-Audio console? Simple: when our
team of obsessive console engineers first began designing Element, they asked broadcast professionals
to describe their ideal mixing console. “Powerful,” they said, “but easy to use, with the capabilities of a
full-up production board.”
So our engineers went to the lab and blended the best ideas from old-school analog consoles with
innovative new technology to produce bullet-proof boards that can actually make shows run smoother
and sound better.
Like all Axia broadcast equipment, Element consoles connect using standards-based Livewire IP-Audio
networking technology, invented by Telos. Using Livewire, broadcasters can easily network studios,
consoles and audio equipment using standard Ethernet. Livewire can carry hundreds of channels of
real-time, uncompressed audio plus synchronized control logic and program-associated data on a single
CAT-6 cable, reducing cost, complexity and studio construction time.
Because Axia networks are intelligent Ethernet-based routing systems, machine logic always follows
source audio. When your operator loads a source to any fader, in any studio, that fader’s controls are
immediately communicating with the source device. Thanks to this scalable network technology,
integrated router control is a standard feature of every Element.
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Talent raves about Element Show Profiles. Each console contains up to 99 storage locations that
operators can use to set, save and recall their favorite settings with the push of a button — audio
sources, fader assignments, monitor settings and more. Show Profiles can also contain talent’s
personalized mic processing and voice EQ settings that load every time they’re on the air — and, in case
a jock gets himself into unfamiliar territory, Element provides a convenient one-key “panic button” that
returns a Show Profile to its default state instantly.
There’s plenty of power under the surface, too. To make sure you have plenty of mixing capacity, Element
features 4 Program buses, plus 4 Aux sends and 2 Aux returns, along with 16 five-channel “Virtual
Mixers” that let you mix multiple audio inputs using virtual faders. More built-in convenience: Every voice
channel has studio-grade Omnia audio processing, including mic compression, de-essing and gating,
plus three-band parametric EQ, which can be set and saved with each Show Profile. Need a headphone
processor for your talent? Element provides that, too, with built-in headphone processing to save the
cost of a separate side-chain.
You’ll also find fully-automatic mix-minuses; one for each fader if needed. Mix-minus settings are saved
for each audio source, so that sources, backfeed and machine logic all load at once. And every fader has a
“Talkback” key to communicate with phone callers, remote talent or other studios using the console mic;
use them singly, or in multiple to communicate with entire groups of locations at once.
Axia’s Livewire Ethernet backbone makes it easy to integrate and control all kinds of different devices
on the same network. And Element puts those controls right on the console, where they’re most useful.
For instance: phone hybrid modules with dedicated faders control Telos talkshow systems. There’s a dial
pad, too, so talent can dial, answer, screen and drop calls without ever taking their eyes – or attention
–off the console. Which translates into smoother, more error-free on-air phone segments. Axia’s IP
Intercom system connects to the Livewire network too, and drop-in Intercom modules for your Element
place multi-station intercom controls right at jocks’ fingertips. Which means that talent can now easily
take broadcast-quality intercom audio directly to air, with only a button-press or two.
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As with all Axia consoles, engineers can administer Element remotely. A password-protected Web server
lets you examine the state of the console and make configuration changes. With SoftSurface companion
software, you can even take direct remote control of Element from your office, home, or anywhere
there’s an Internet connection.
There’s more to building a great board than just features, of course. Consoles have to perform flawlessly
24/7, 365 days-a year, for years at a time. So Element is fabricated from thick, machined aluminum
extrusions — rigid and RF-immune. Power supplies are hardened for reliable, continuous uptime, and
fanless for silent in-studio operation. Modules can be hot-swapped. Silky-smooth conductive-plastic
faders actuate from the side, so dirt can’t get in. High-end optical rotary encoders mean no wipers to
get dirty or wear out. And our avionics-grade switches, with LED lighting, have been tested to withstand
more than five million operations.
Some folks have said that Element consoles are over-engineered. To which we say, “thank you”! Not
everyone appreciates this kind of attention to detail, but if you’re one who seeks out and appreciates
excellence wherever you may find it... Element just may be the answer you’ve been looking for.
Your station is customized to your listeners. Shouldn’t your console be customized to your talent?
Mix and match a variety of Element module types with enhanced features to suit your station’s
operational needs. Like integrated controls for phones, codecs and intercoms, EQ modules designed
to speed off-air production, even motorized faders for remote control or integration with your delivery
system. Choice is good!
4-Fader Module
The 4-Fader module is the heart of any Element. Use it for any source: line, mic,
hybrid, phone or codec source. Comes in standard and motorized-fader versions
for use with automation systems or other moving-fader applications.
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Monitor Modules
The other basic module every console needs is the Monitor module. Element
offers two types.
The Expert Monitor/Navigation module shown here has extended monitor,
headphone and preview controls, plus a numeric entry/dialpad that can be used
with Element phone modules, plus convenient profanity delay controls that can
be linked to your delay unit.
For studios where expert monitor controls are not needed, the Standard
Monitor/Navigaion module is a space-saving design that incorporates two
faders in addition to the numeric entry/dial pad and basic Monitor/
Headphone controls.
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Call Controller Module
The Call Controller module has two faders plus integrated line switching
controls with Status Symbols, for on-console control of advanced Telos
broadcast phone systems. Available in standard and motorized-fader
versions.
Switch Modules
Two available styles of programmable switch modules work with Axia
PathfinderPC routing control tools. They make it easy to put custom routing
salvos or simple machine logic right at talent’s fingertips.
Economical Film-Cap switch modules are perfect for giving talent access
to often-used machine-control or GPIO-triggered routing commands. LED
button backlights can be individually changed to any of 8 colors.
Need to give operators more complex control of routing functions?
SmartSwitch modules feature dynamic, backlit LCD displays. Button
functions, colors and even text can be programmed to change in response
to user input using Axia PathfinderPC software. Construct custom routing
salvos, cascading machine-logic command sets, or other complex routing
operations.
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IP Intercom Modules
Element consoles come equipped with a sophisticated Talkback system that
allows board ops to communicate directly with remote talent via individual
Talkback channels. But when larger facilities require even more powerful
communication capabilities, these 10 and 20-station intercom modules,
part of the Axia IP Intercom system, put broadcast intercom controls right in
the console. Station presets and GPIO functions for both types of modules
are programmed using any standard Web browser.
10 and 20-station OLED Intercom modules feature high-resolution
programmable OLED displays that indicate assigned stations. The
10-Station Filmcap intercom module has 10 LED-lit film-cap buttons for
economical on-console IP Intercom integration.
Mixing Engines
Element consoles give you choices at every turn, and mixing engine platforms are no exception. You
can build your Axia network using a la carte components – an Element control surface with a powerful
Linux-based StudioEngine and separate xNode audio interfaces – or you can choose the PowerStation
integrated console engine, an all-in-one powerhouse with audio I/O, DSP mixing engine and integrated
zero-configuration network switch.
StudioEngine
Pair your Element with Axia StudioEngine, an extremely powerful mixing and processing device based
on a blazingly-fast Intel processor. Each StudioEngine is fanless, has dual-redundant field-replaceable
modular power supplies, and has so much CPU power it can outperform the very largest digital or
router-based consoles. StudioEngine has multiple simultaneous inputs, outputs, mix-minus feeds,
monitor signals, EQ and voice processing; it’s the power behind state-of-the-art broadcast studios from
New York to Tokyo.
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PowerStation
PowerStation is what we Axians refer to as an “integrated console engine”, an all-in-one titan that
makes it easier than ever to install Axia studios with Element consoles. Inside that ruggedly handsome
case you’ll find a super-powered DSP mixing engine, husky power supply sourced from telecom gear
designed for harsh environments, plenty of built-in digital, analog and mic I/O, plus EQ, voice processing
— and even a custom, built-for-broadcast Ethernet switch with Gigabit connectivity.
SPECIFICATIONS
Microphone Preamplifiers
• Source Impedance: 150 Ohms
• Input Impedance: 4 k Ohms minimum, balanced
• Nominal Level Range: Adjustable, -75 dBu to -20 dBu
• Input Headroom: >20 dB above nominal input
• Output Level: +4 dBu, nominal
Analog Line Inputs
• Input Impedance: >40 k Ohms, balanced
• Nominal Level Range: Selectable, +4 dBu or -10dBv
• Input Headroom: 20 dB above nominal input
Analog Line Outputs
• Output Source Impedance: <50 Ohms balanced
• Output Load Impedance: 600 Ohms, minimum
• Nominal Output Level: +4 dBu
• Maximum Output Level: +24 dBu
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Digital Audio Inputs and Outputs
• Reference Level: +4 dBu (-20 dB FSD)
• Impedance: 110 Ohms, 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
Equivalent Input Noise
• Microphone Preamp: -128 dBu, 150 ohm source, reference -50 dBu input level
Total Harmonic Distortion + Noise
• Mic Pre Input to Analog Line Output: <0.005%, 1 kHz, -38 dBu input, +18 dBu output
• 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
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Crosstalk Isolation, Stereo Separation and CMRR
• Analog Line channel to channel isolation: 90 dB isolation minimum, 20 Hz to 20 kH
• Microphone channel to channel isolation: 80 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
• Microphone Input CMRR: >55 dB, 20 Hz to 20 kHz
Audio Processing
Equalizer
• Frequency Bands: 20Hz to 320Hz, 125Hz to 2KHz, 1.25KHz to 20KHz.
• Cut/Boost range on each band: -25dB to +15dB.
• Q-factor: Automatic - bandwidth varies based on amount of cut or boost.
Compressor
• Threshold: -30dB to 0dB Ratio: 1:1 to 16:1
• Post-processor Trim Level: Adjustable from -20dB to +20dB
Expander/Noise Gate
• Threshold: -50dB to 0dB Ratio: -30dB to 0dB
De-esser
• Threshold: -20dB to 0dB Ratio: 1:1 to 8:1
Power Supply AC Input, StudioEngine
• Auto-sensing, field-replaceable modular supply, 90VAC to 240VAC, 50 Hz to 60 Hz, IEC receptacle,
internal fuse
• Power consumption: 100 Watts
Power Supply AC Input, Element Power Supply/GPIO
• Auto-sensing supply, 90VAC to 240VAC, 50 Hz to 60 Hz, IEC receptacle, internal fuse
• Power consumption: 150 Watts
Power Supply AC Input, PowerStation Aux & Main
• Auto-sensing supply, 90VAC to 240VAC, 50 Hz to 60 Hz, IEC receptacle, internal fuse
• Power consumption: 500 Watts
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Axia Console Power Supply
• Add redundant power to PowerStation main without additional IO.
• Single-cable connection to PowerStation main provides backup power with automatic switching.
• Auto-sensing power supply, 90VAC to 240VAC, 50 Hz to 60 Hz.
• Power consumption: 250 Watts.
Operating Temperatures
• -10 degrees C to +40 degrees C, <90% humidity, no condensation
Regulatory
North America: FCC and CE tested and compliant, power supply is UL approved.
Europe: Complies with the European Union Directive 2002/95/EC on the restriction of the use of certain
hazardous substances in electrical and electronic equipment (RoHS), as amended by Commission
Decisions 2005/618/EC, 2005/717/ EC, 2005/747/EC (RoHS Directive), and WEEE.
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SoftSurface
Software for Axia® Fusion™ and
Element® Consoles
OVERVIEW
SoftSurface Virtual Console software for Windows gives you powerful real-time control of your Axia
Fusion or Element mixing console from home, office, or anywhere an Internet connection is available.
Take direct remote control of your console, or, match SoftSurface directly to an Axia mixing engine to
create a “virtual console” without a physical mixing surface. SoftSurface makes an ideal companion
for existing consoles and it’s also the perfect audio mixing solution for limited-space locations.
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FEATURES
• Gives full remote control of Fusion or Element consoles paired with Axia StudioEngine mixing engines.
• Pair directly with an Axia StudioEngine or PowerStation® to create a standalone “soft” console without
a physical mixing surface. Supports from 4 to 48 faders in this mode.
• NTP-capable on-screen time of day clock/calendar.
• On-screen count-up event timer.
• Supports up to four Show Profiles console “snapshots” for instant recall of frequently-used
configurations.
• Control all four program buses and all auxiliary mix buses.
• Remote control of mic compression, de-essing and expansion capabilities.
• On-screen control of per-source three-band parametric EQ.
• Excellent IFB Talkback capabilities let operators talk to other studios, external remote feeds, phone
callers or any other source with its own backfeed.
• Full control of GPIO functions.
• When paired with Axia consoles equipped with motorized faders, physical fader position automatically
mirrors that of the “virtual” SoftSurface fader.
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IN DEPTH
Control at the Click of a Mouse.
You asked for a way to remotely control your console. Axia heard you! Meet SoftSurface, the audio
mixing application for Windows.
You can use SoftSurface two ways. As a remote control, it gives powerful real-time control of premium
Axia mixing consoles, utilizing an Axia StudioEngine or PowerStation that’s connected to a Livewire®
network. It’s perfect for remote diagnostics or off-site operation of a mixing console from remotes,
transmitter sites — even from home, via an Internet gateway.
As a virtual console, SoftSurface combines with an Axia StudioEngine or PowerStation to create a “soft”
mixing surface. It’s perfect for those limited-space situations where there’s no room for a real console.
With SoftSurface, if you’ve got a Windows laptop and an IP connection, you’re good to go.
Now, when we say SoftSurface lets you control a console, we don’t mean a wimpy console. You get all
the functionality and features of Axia’s extremely popular Element and Fusion modular mixing surfaces,
the boards at the heart of thousands of superb broadcast facilities around the world.
SoftSurface opens up new dimensions of creative applications for broadcasters. Remote broadcasts get
easier: your talent can take a tablet with SoftSurface, a USB mic and a Telos® Z/IP ONE IP Codec into
the field, link up with the Element console at your studio, and have its entire suite of capabilities at their
fingertips — leave those CDs and MP3 players at home.
Or, pair SoftSurface with a StudioEngine or PowerStation mixing engine for a “virtual console” installation
in personal studios, or areas where space restrictions don’t permit a physical control surface.
The SoftSurface display is divided into a virtual mixing surface and a control section. The mixing section’s
onscreen width varies based on the number of channels you wish to display, while the context-sensitive
control section is fixed in size, and navigation is via a series of intuitive tabs.
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The Main Monitor tab provides control of the monitor speakers, operator headphones, external monitor
speakers, and preview volume.
The Show Profiles tab reveals the profiles that have been configured and allows the user to select a
show and load it. As few as one or as many as 99 Show Profiles are supported.
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Monitor Options provides control over dimming values and control of the channel feeding the monitors
and headphone.
The Meter Options tab provides control over the presentation of the meters, including choice of metering
ballistics styles: VU, BBC and DIN-style PPM, EBU Digital and Nordic.
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Auxiliary Send and Return provides control of the final AUX send mix, as well as two auxiliary returns, as
defined in the Show Profile. The returns can be assigned to a program bus. These options are defined in
a Show Profile and controlled as needed by the user. Individual sources can be assigned to feed one or all
AUX Send buses from the Channel Options Aux Send screen.
EQ curves are adjustable for each audio source. EQ can be adjusted on the fly, or saved as part of a
source’s Source Profile and automatically recalled whenever that source is loaded to a fader.
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Mic, Codec and Phone sources can be sweetened with Voice Dynamics from Omnia®. Expansion,
Compression and De-Essing are part of the toolkit; Like EQ, dynamics settings can be adjusted at will or
preset and saved with sources for automatic recall.
SoftSurface’s signal processing toolkit is completed with comprehensive Pan, Summing and Phase
control, available on a per-source basis.
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SPECIFICATIONS
Hardware Requirements
• A PC with Microsoft™ Windows XP or higher operating system installed.
• A display screen with a minimum resolution of 1280x1024 pixels.
• A mouse, touch-screen, or other suitable pointing device.
• A 100BASE-T LAN connection with a static IP address.
• 20Mb free disk space.
Audio Processing
Equalizer
• Frequency Bands: 20Hz to 320Hz, 125Hz to 2KHz, 1.25KHz to 20KHz.
• Cut/Boost range on each band: -25dB to +15dB.
• Q-factor: Automatic - bandwidth varies based on amount of cut or boost.
Compressor
• Threshold: -30dB to 0dB Ratio: 1:1 to 16:1
• Post-processor Trim Level: Adjustable from -20dB to +20dB
Expander/Noise Gate
• Threshold: -50dB to 0dB Ratio: -30dB to 0dB
De-esser
• Threshold: -20dB to 0dB Ratio: 1:1 to 8:1
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AXIA | IQ
iQ
The Smarter IP Console
OVERVIEW
The Axia® iQ console system can be used to build custom consoles of sizes from 8 to 24 faders. A basic
system consists of one iQ 8-Fader Main Frame and one QOR.32 integrated console engine, a DSPbased mixing engine which also incorporates analog and digital audio I/O, GPIO and a custom, zeroconfiguration Ethernet switch. Faders and control capabilities can be expanded by adding one or more
iQ Expansion Frames (up to a maximum of 3 frames per console installation). iQ console frames may
be placed on top of desk surface, or mounted drop-in style. Multiple frames may be physically joined if
desired.
iQ operates as a standalone console, but can also connect to Axia networks. The iQ mixing surface plugs
into the QOR.32 engine using a single cable. Setup couldn’t be simpler: connect the iQ control surface
to the QOR.32, add audio inputs using CAT-5, perform some fast Web-based configuration, and your iQ
system is ready to broadcast. It really is that simple!
iQ features 3 dedicated stereo Program buses, plus a stereo Utility bus that can be used for phone
calls, off-air recording, or as a fourth Program bus. Automatic mix-minus is provided on each fader, plus
talkback functions, one-button off-air Record Mode, Show Profile functions for instant recall of up to 4
pre-defined console “snapshots”, high-resolution OLED program meters switchable between VU and
PPM metering styles, OLED option and source name displays on each fader strip, Studio and Control
Room monitor controls. Add redundant power to your iQ system with the Axia Console Power Supply.
This single-cable connection to the QOR.32 console engine provides backup power with automatic
switching.
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FEATURES
• Configurable from 8 to 24 faders, each with instant access to any source.
• Proven surface-and-core architecture separates control from mixing processes. No audio passes
directly through iQ; all mixing and processing is performed in the QOR.32 Integrated Console Engine –
so studio “accidents” don’t turn into off-air events.
• Software upgrade for QOR.32 integrated console makes iQ AoIP console AES67-compliant.
• Assign any type of source to any channel with a twist of the Options knob.
• Four main stereo outputs (Program-1 through Program-4).
• Built-in three-band per-source EQ.
• Alpha-numeric OLED displays below each fader always show the current audio source, and, when the
Options knob is pressed, offer fast adjustment of fader gain trim, voice EQ, pan and balance, phase
correction and other features without panel clutter or intimidating controls.
• Channel-input confidence meters assure talent of audio presence before taking sources to air.
• Each fader’s context-sensitive Soft key can be used to activate talkback, start delivery system events,
or perform other special functions.
• Every fader has a stereo Preview (“cue”) function, with a unique interlock system for fast cuing of
multiple sources.
• Smooth, long-life 100mm. conductive-plastic faders resist dirt and contamination.
• Reconfigurable CR monitor section with direct-selection of Program buses and reassignable buttons
that allow instant monitoring of external sources.
• An additional monitor section provides monitor volume, source selection and talkback controls for an
associated air studio.
• Flexible talkback system lets board op talk to studio guests or any Phone or Codec source with an
associated backfeed.
• Up to 8 automatic mix-minuses may be used simultaneously for phones, remote talent, etc.
• Unique Record Mode enables one-button setup of record mixes for phone bits or off-air interviews.
• High-resolution OLED displays provide responsive, readable VU or PPM metering styles. Displays can
be switched to display 2, 3 or 4 meters at once.
• Precision event timer that can be operated manually or triggered by starting pre-selected sources.
• Time-of-day clock can be synchronized to network time using NTP.
• Four custom Show Profile “snapshots” can be saved to instantly recall frequently-used console setups –
useful to quickly prepare for interview segments, music-intensive programming, call-in talk shows, etc.
• All functions can be accessed remotely for configuration, management and diagnostic purposes using
any standard Web browser.
• Multiple iQ frames can be joined to produce a single, large control surface, or operated separately if
desired to suit studio design.
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• Optional Telco Expansion frame provides direct, on-the-console control of matching Telos® iQ6 sixline telephone system, or other Telos talkshow systems. High-resolution OLED displays use exclusive
Telos Status Symbols for instant call status information. Includes a Dump key to trigger user-supplied
profanity delay unit using GPIO closures.
• Easy-to-deploy QOR.32 integrated console engine includes console CPU and power supply, DSP
mixing engine, custom Ethernet switch with 6 Livewire® ports and 2 Gigabit ports for studio
networking, 16 analog inputs and 8 analog outputs, 2 AES inputs and 2 AES outputs, 4 Mic inputs
with switchable Phantom power, and 8 GPIO ports for machine control. I/O can be expanded using
Telos Alliance® xNodes.
• Integrated zero-configuration network switch is custom-designed for broadcasting — no switch
setup required.
• QOR.32’s built-in Ethernet switch supports Simple Networking, allowing up to 4 iQ consoles to be
daisy-chained without the need for a separate core switch.
• Fan-free, convection-cooled power supply for noiseless in-studio operation.
• Optional backup Axia Console Power Supply with automatic failover for complete peace of mind.
• Configurable network gateway allows loading of networked as well as local audio sources while
simultaneously exporting audio streams for network use elsewhere. Gateway can be configured for
12-in, 4-out or 8-in, 8-out modes.
IN DEPTH
Control at the Click of a Mouse.
Easy Installation. Fast Configuration.
Intuitive Operation.
For today’s broadcast engineer, there aren’t enough hours in the day. You’re looking for a console that
makes the most of your resources. One that installs quickly, with a minimum of fuss. One that works
smart, with features that help talent to do smoother, more error-free shows. One that’s perfectly happy
in a standalone studio — but that also connects quickly and easily to a larger studio network.
iQ is the console you’re looking for. More than just a pretty face, iQ is a broadcast console with mixing
engine, analog and AES audio I/O, Livewire audio connections, machine-control logic and a zeroconfiguration built-for-broadcast Ethernet switch, all rolled into one easy-to-deploy package. Connect
the iQ control surface to the QOR.32 integrated console engine with just one cable. Then add audio
inputs using CAT-5, perform some fast Web-based configuration and, presto! your new iQ console
is ready to broadcast. Optional Axia Console Backup Power Supply adds redundant power to your iQ
system for complete peace of mind.
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Thanks to all those built-in goodies, iQ is the perfect self-contained, standalone console for an individual
studio. But should you wish to expand and network with other studios, iQ can grow with you. Simple
Networking lets you daisy-chain up to four QOR.32 engines without the need for an external Ethernet
switch. You can add iQ expansion frames to create consoles as large as 24 faders. Other optional frames
add control for Telos telephone systems and GPIO routing functions to the console.
More smart stuff: iQ remembers. Four Show Profile memory positions let you set, save and recall
snapshots of console settings for later use. High-resolution Organic LED meters (bright, high-resolution
displays that are bright and legible, even under direct lighting) offer switchable VU or PPM metering
styles, and the ability to meter two, three, or all four buses at once.
There are also OLED displays on every fader that provide source assignments, pan & balance settings,
fader options and more — which means no additional computer monitors or mice to clutter up your
studio. The display can also work with the Soft Keys just below to trigger GPIO events, step automation
events, and adjust source input options.
iQ saves your studio furniture, too. Its desktop design lets you place it atop any solid surface — no
templates to decipher or countertops to cut (unless you really want to). Since iQ only requires a single
cable to connect control surface to mixing engine, even cable access holes can be small and unobtrusive.
And iQ lets you choose between freestanding or contiguous console designs: you can easily join iQ
expansion frames into one unit, or leave them separate to deploy a split-console design.
Like all Axia consoles, iQ is over-engineered for long life. It’s built with sturdy, premium materials, to
withstand even the beatings a weekend overnight jock can give. It’s got sturdy, machined aluminum
frame construction, LED button lighting, long-life conductive-plastic faders, and anodized – not painted!
– surfaces with laser-etched markings that can’t ever rub off. But the most clever thing about iQ might
just be its price. A 16-fader iQ costs about half what you’d expect to pay for a console with all these
features. Now that’s pretty smart, don’t you think?
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iQ System Components
Like all Axia systems, iQ is customizable and scalable. The QOR.32 integrated console engine contains
the console’s mix engine, CPU, power supply and 32 audio I/O connections, and supports console sizes
from 8 to 24 faders. Start with an eight-fader iQ Main Frame, then add expansion frames with more
faders and capabilities to tailor iQ to your studio’s needs. Gigabit Ethernet lets you connect to a larger
Axia network; Simple Networking lets you daisy-chain up to four QOR.32 without the need for an
external Ethernet switch.
iQ Main Frame
The heart of your iQ console; can be installed as a standalone console or connected to an Axia studio
network. Has three dedicated stereo Program buses, plus a stereo utility bus that can be used for phone
calls, off-air recording, or as a fourth Program bus, eight faders, automatic per-fader mix-minus, highrez OLED program meters and channel displays, Studio and Control Room monitor controls and an
integrated Talkback system. For bigger consoles, add one or two iQ expansion frames to build boards of
up to 24 faders. Flexible mounting system allows desktop, drop-in and even rack-mounted operation.
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8-Fader Expansion Frame
The iQ 8-Fader frame doubles the size of your iQ instantly. It’s simple to expand the capacity of iQ
consoles, even after they’ve been in service, so you can easily grow your iQ system; expansion frames
plug right into the QOR.32 integrated console engine. Like all iQ frames, the 8-Fader expansion comes
equipped with Axia’s rugged, anodized machined-aluminum surface, conductive-plastic faders, aircraftquality switches and LED button lighting. Can be physically joined to Main Frame or left separate.
6-Fader Expansion Frame with User Keys
Put machine control and GPIO-triggered routing commands at your operators’ fingertips with this iQ
expansion frame. In addition to the six additional faders, 10 User Keys can be software-mapped to
control audio delivery systems, send contact closures or route GPIO commands to studio devices.
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6-Fader Telco Expansion Frame
Puts integrated phone system control right where it belongs: on the console, to help eliminate
distractions and errors. Along with six silky-smooth conductive-plastic faders, this frame includes onthe-board hybrid controls for the matching Telos iQ6 six-line telephone hybrid (it works with other Telos
phone systems, too). The learning curve is low: exclusive Telos Status Symbols readouts on sharp-as-atack OLED displays, along with familiar twin hybrid controls, make easy work of busy call-in segments.
iQ6 6-Line Telco Gateway
The iQ6 broadcast phone system was custom-designed by the phone experts at Telos specifically for iQ
consoles. It works with the hybrid controls built into your iQ’s Telco expansion frame (and with Telos
VSet6 phone controllers). Connect it to the QOR.32 console engine with a CAT-6 cable, plug in your
phone lines, and start taking calls.
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QOR.32 Integrated Console Engine
The QOR.32 integrated console engine is a DSP-based mixing engine with onboard I/O, GPIO, console
power supply and custom-built, configuration-free Ethernet switch. You’ll find plenty of I/O, including
4 mic inputs with selectable Phantom power, 16 analog inputs, 2 AES/EBU inputs, 8 Analog outputs, 2
AES/EBU outputs, 8 GPIO machine-control logic ports (each with 5 opto-isolated inputs and 5 outputs),
and that powerful integrated Ethernet switch with 6 Livewire 100BASE-T ports (4 with PoE), 2 Gigabit
ports (RJ-45 & SFP), and 4 CANBus ports for console expansion. Sure, that’s plenty of I/O, but if you need
more you can instantly add it just by plugging in Telos Alliance xNodes. QOR.32 is convection-cooled for
utterly silent, fan-free operation.
Axia Console Backup Power Supply
Add redundant power to your iQ system! Single-cable connection to QOR.32 console engine provides
backup power with automatic switching. Auto-sensing power supply, 90VAC to 240VAC, 50 Hz to 60 Hz.
250 Watts. Rackmount, 2RU.
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SPECIFICATIONS
QOR.32 Connections
• Microphone Inputs: 4x balanced XLR-F, with selectable Phantom power
• Analog Inputs: 16x RJ-45, StudioHub+ standard.
• Analog Outputs: 8x RJ-45, StudioHub+ standard.
• AES/EBU Inputs: 2x RJ-45, StudioHub+ standard.
• AES/EBU Outputs: 2x RJ-45, StudioHub+ standard.
• GPIO: 8x DB-15
• Livewire:
• 4x 100BASE-T with PoE, RJ-45
• 2x 100BASE-T, RJ-45
• 2x 1000BASE-T, RJ-45
• 2x Gigabit, SFP (Small Form Pluggable)
• Console Frame Connections: 3x, 6-pin “latch and lock” style
• Accessory Connections: 1x, 6-pin “latch and lock” style
Microphone Preamplifiers
• Source Impedance: 150 Ohms
• Input Impedance: 4 k Ohms minimum, balanced
• Nominal Level Range: Adjustable, -75 dBu to -20 dBu
• Input Headroom: >20 dB above nominal input
• Output Level: +4 dBu, nominal
Analog Line Inputs
• Input Impedance: 20 k Ohms
• Nominal Level Range: Selectable, +4 dBu or -10dBv
• Input Headroom: 20 dB above nominal input
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Analog Line Outputs
• 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, 20 kHz to 216kHz 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: 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: 125 dB
Equivalent Input Noise
• Microphone Preamp: -128 dBu, 150 Ohm source, reference -50 dBu input level
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Total Harmonic Distortion + Noise
• Mic Pre Input to Analog Line Output: <0.005%, 1 kHz, -38 dBu input, +18 dBu output
• 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 kH
• Microphone channel to channel isolation: 80 dB isolation minimum, 20 Hz to 20 kHz
• Analog Line Stereo separation: 85 dB isolation minimum, 20Hz to 20 kHz
• Analog Line Input CMRR: >50 dB, 20 Hz to 20 kHz
Microphone Input CMRR: >50 dB, 20 Hz to 20 kHz
Audio Processing
• Mic Equalizer (applicable to up to 6 faders)
• Frequency Bands: 20Hz to 320Hz, 125Hz to 2KHz, 1.25KHz to 20KHz.
• Cut/Boost range on each band: -25dB to +15dB.
• Q-factor: Automatic - bandwidth varies based on amount of cut or boost.
Power Supply AC Input, QOR.32 with iQ Console
• Auto-sensing supply, 90VAC to 240VAC, 50 Hz to 60 Hz, IEC receptacle, internal fuse
• Power consumption: 100 Watts
Operating Temperatures
• -10 degrees C to +40 degrees C, <90% humidity, no condensation
Dimensions
• iQ Main Frame 20.5” x 19” x 4.5” (desktop to meter bridge)
• iQ Expansion Frames 17.5” x 18.25” x 3” (desktop to tallest control)
Regulatory
North America: FCC and CE tested and compliant, power supply is UL approved.
Europe: Complies with the European Union Directive 2002/95/EC on the restriction of the use of certain
hazardous substances in electrical and electronic equipment (RoHS), as amended by Commission
Decisions 2005/618/EC, 2005/717/ EC, 2005/747/EC (RoHS Directive), and WEEE.
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AXIA | RADIUS
Radius
Throw Your Budget A Curve
OVERVIEW
Radius is an all-in-one console system designed for small standalone or networked studios, where
no more than eight faders are needed. Like all Axia® consoles, it’s easy to deploy: each Radius control
surface is powered by a burly QOR.16 integrated console engine with DSP-powered mixing engine,
analog and digital audio I/O, custom Ethernet switch and GPIO ports. The Radius surface connects to the
QOR.16 engine with a single CANBus cable.
Radius includes 4 stereo Program buses — 3 dedicated Program, Audition and Utility mixing outputs;
the fourth a stereo Utility bus for recording phone callers or other off-air bits. The fourth bus may also
be used as an additional Program bus. Automatic mix-minus is provided on each fader, plus talkback
functions, one-button off-air Record Mode, Show Profile instant recall of up to 4 pre-defined console
“snapshots”, LED bar-graph program meters switchable between VU and PPM meter styles, highresolution OLED option displays on each fader, and Studio and Control Room monitor controls. Radius
can be placed on top of desk surfaces, mounted drop-in style, or rack-mounted using included hardware.
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FEATURES
• 8 faders, each with instant access to any source. Assign any type of source to any channel with a
simple twist of the Options knob.
• Proven surface-and-core architecture separates control from mixing processes. No audio passes
directly through Radius; all mixing and processing is performed in the QOR.16 Integrated Console
Engine – so studio “accidents” don’t turn into off-air events.
• Four main stereo outputs (Program-1 through Program-4).
• Software upgrade for QOR.16 integrated console makes Radius AoIP console AES67-compliant.
• Built-in three-band per-source EQ.
• Alpha-numeric OLED displays below each fader always show the current audio source, and, when the
Options knob is pressed, offer fast adjustment of fader gain trim, voice EQ, pan and balance, phase
correction and other features without panel clutter or intimidating controls.
• Channel-input confidence meters assure talent of audio presence before taking sources to air.
• Each fader’s context-sensitive Soft key can be used to activate talkback, start delivery system events,
or perform other special functions.
• Every fader has a stereo Preview (“cue”) function, with a unique interlock system for fast cuing of
multiple sources.
• Smooth, long-life 100mm. conductive-plastic faders resist dirt and contamination.
• Reconfigurable CR monitor section with direct-selection of Program buses and reassignable buttons
that allow instant monitoring of external sources.
• An additional monitor section provides monitor volume, source selection and talkback controls for an
associated air studio.
• Flexible talkback system lets board op talk to studio guests or any Phone or Codec source with an
associated backfeed.
• Up to 8 automatic mix-minuses may be used simultaneously for phones, remote talent, etc.
• Unique Record Mode enables one-button setup of record mixes for phone bits or off-air interviews.
• Bright, readable bar-graph displays provide responsive, readable VU or PPM metering styles.
Switchable displays allow metering any Program bus or monitor selection.
• Meter-bridge display includes a precision event timer that may be operated manually or triggered
by starting preselected sources, and a time-of-day clock that can be synchronized to network time
using NTP.
• Four custom Show Profile “snapshots” can be saved to instantly recall frequently-used console
setups – useful to quickly prepare for interview segments, music-intensive programming, call-in
talk shows, etc.
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• All functions can be accessed remotely for configuration, management and diagnostic purposes using
any standard Web browser.
• Radius surface is field-convertible to rack-mounted operation. Special faders provide smooth
operation, yet hold their positions in vertical orientation.
• Easy-to-deploy QOR.16 integrated console engine includes console CPU and power supply, DSP
mixing engine, custom Ethernet switch with 6 Livewire® ports and 2 Gigabit ports for studio
networking, 8 analog inputs and 4 analog outputs, 1 AES input and 1 AES output, 2 Mic inputs
with switchable Phantom power, and 4 GPIO ports for machine control. I/O can be expanded using
Telos Alliance® xNodes.
• Integrated zero-configuration network switch is custom-designed for broadcasting — no switch
setup required.
• QOR.16’s built-in Ethernet switch supports Simple Networking, allowing up to 4 iQ consoles to be
daisy-chained without the need for a separate core switch.
• Fan-free, convection-cooled power supply for noiseless in-studio operation.
• Configurable network gateway allows loading of networked as well as local audio sources while
simultaneously exporting audio streams for network use elsewhere. Gateway can be configured for
12-in, 4-out or 8-in, 8-out modes.
IN DEPTH
Spend Less. Get More.
“You get what you pay for,” as the saying goes. But sometimes, you actually get less. For example, you’ve
probably noticed how “affordable” radio consoles are usually missing important features and capabilities.
Trying to do a radio show with a board like that is like trying to open a can with a spoon: you might
succeed eventually, but you sure won’t enjoy it.
At Axia, we’re broadcasters too, through and through. And we believe that having a reasonable
equipment budget shouldn’t mean being forced to settle for something less than you deserve. We’ve
decided you should get more than you pay for — much more. Which is why we designed Radius, the
IP console that proves you can have your cake and eat it, too. While some console companies try to
see how much they can take out of a console to meet a price point, Radius was designed in exactly the
opposite way: we challenged ourselves to see just how many features and capabilities we could pack in,
while still meeting your budget requirements.
Radius is the easiest AoIP console ever. Just connect the 8-fader mixing surface to the QOR.16
integrated console engine, plug in your sources and power, and you’re ready to make great radio.
Because it’s so compact, Radius is the perfect standalone console, but Gigabit ports on its QOR.16
integrated console engine let you connect it to other studios too. Radius’ network gateway lets you load
up to 12 audio sources from anywhere in your Livewire network, while simultaneously sending your
locally produced streams back out to the net.
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Radius is loaded with features you’d expect to pay much more for. You’ll find three stereo Program
buses, and a stereo utility bus that can be used for recording phone calls and off-air bits (or as a fourth
Program bus). Automatic mix-minus for every phone caller and remote talent means never having to
fiddle with making a manual backfeed. Bright multi-segment LED meters are switchable between VU
and PPM styles. High-resolution OLED displays for each fader show source assignments, audio options
and more. And Show Profiles that you can program to instantly load talent’s most frequently-used
console configurations.
Like all Axia consoles, Radius is built for long-lasting reliability, ready to stand up to anything your
operators throw at it, with an EM-tight steel frame, anodized machined aluminum work surface with
etched markings that can never rub off, silky-smooth conductive-plastic faders, aircraft-quality
switches and rotary controls, and integrated clock/event timer. There are even monitor source and
volume controls for an associated studio — something you’d expect to find only in bigger consoles
costing much more.
There are also OLED displays on every fader that provide source assignments, pan & balance settings,
fader options and more — which means no additional computer monitors or mice to clutter up your
studio. The display can also work with the Soft Keys just below to trigger GPIO events, step automation
events, and adjust source input options.
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Like its big brother, iQ, Radius is designed to sit atop any solid surface — no templates to decipher or
countertops to cut (unless you really want to). A single cable connects it to the QOR.16 mixing engine.
Also like iQ, Radius is built with premium materials like a machined aluminum frame construction, LED
button lighting, long-life conductive-plastic faders, and anodized – not painted! – surfaces with laseretched markings that can’t ever rub off.
Audio I/O, GPIO, console CPU, super-duty power supply, and even a network switch are all built into the
QOR.16. Just plug in your mics, CD players, codecs, profanity delays, whatever. There are 16 audio I/O
ports: two Mic inputs with switchable Phantom power, eight analog inputs and four analog outputs,
and one AES/EBU input and output. QOR.16 also has four GPIO logic ports for machine control of studio
peripherals, six 100BASE-T ports for Livewire devices, and two Gigabit ports with SFP for connection
to the outside world. For more I/O, just add Telos Alliance xNode interfaces. And you can daisy chain as
many as four QOR engines without the need for an external Ethernet switch, making installation even
more economical.
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SPECIFICATIONS
QOR.16 Connections
• Microphone Inputs: 2x balanced XLR-F, with selectable Phantom power
• Analog Inputs: 8x RJ-45, StudioHub+ standard.
• Analog Outputs: 4x RJ-45, StudioHub+ standard.
• AES/EBU Inputs: 1x RJ-45, StudioHub+ standard.
• AES/EBU Outputs: 1x RJ-45, StudioHub+ standard.
• GPIO: 4x DB-15
• Livewire:
• 4x 100BASE-T with PoE, RJ-45
• 2x 100BASE-T, RJ-45
• 2x 1000BASE-T, RJ-45
• 2x Gigabit, SFP (Small Form Pluggable)
• Console Frame Connections: 1x, 6-pin “latch and lock” style
• Accessory Connections: 1x, 6-pin “latch and lock” style
Microphone Preamplifiers
• Source Impedance: 150 Ohms
• Input Impedance: 4 k Ohms minimum, balanced
• Nominal Level Range: Adjustable, -75 dBu to -20 dBu
• Input Headroom: >20 dB above nominal input
• Output Level: +4 dBu, nominal
Analog Line Inputs
• Input Impedance: 20 k Ohms
• Nominal Level Range: Selectable, +4 dBu or -10dBv
• Input Headroom: 20 dB above nominal input
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Analog Line Outputs
• 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, 20 kHz to 216kHz 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: 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: 125 dB
Equivalent Input Noise
• Microphone Preamp: -128 dBu, 150 Ohm source, reference -50 dBu input level
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Total Harmonic Distortion + Noise
• Mic Pre Input to Analog Line Output: <0.005%, 1 kHz, -38 dBu input, +18 dBu output
• 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 kH
• Microphone channel to channel isolation: 80 dB isolation minimum, 20 Hz to 20 kHz
• Analog Line Stereo separation: 85 dB isolation minimum, 20Hz to 20 kHz
• Analog Line Input CMRR: >50 dB, 20 Hz to 20 kHz
• Microphone Input CMRR: >50 dB, 20 Hz to 20 kHz
Audio Processing
• Mic Equalizer (applicable to up to 6 faders)
• Frequency Bands: 20Hz to 320Hz, 125Hz to 2KHz, 1.25KHz to 20KHz.
• Cut/Boost range on each band: -25dB to +15dB.
• Q-factor: Automatic - bandwidth varies based on amount of cut or boost.
Power Supply AC Input, QOR.16 with Radius Console
• Auto-sensing supply, 90VAC to 240VAC, 50 Hz to 60 Hz, IEC receptacle, internal fuse
• Power consumption: 100 Watts
Operating Temperatures
• -10 degrees C to +40 degrees C, <90% humidity, no condensation
Dimensions
• 20.5” x 19” x 4.5” (desktop to meter bridge)
Regulatory
North America: FCC and CE tested and compliant, power supply is UL approved.
Europe: Complies with the European Union Directive 2002/95/EC on the restriction of the use of certain
hazardous substances in electrical and electronic equipment (RoHS), as amended by Commission
Decisions 2005/618/EC, 2005/717/ EC, 2005/747/EC (RoHS Directive), and WEEE.
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AXIA | RAQ
RAQ
Rack-mount IP Console
OVERVIEW
Six-fader Axia® RAQ console provides a convenient way to add a physical mixing surface nearly
anywhere, no matter how space-limited. RAQ has six rotary faders with OLED channel options displays,
two stereo mixing buses and Preview (cue) bus, a high-resolution OLED meter display with switchable
VU / PPM ballistics, and monitor / headphone controls for auditioning of Program buses or two
assignable External monitor source selections.
RAQ is built for heavy duty work. Aircraft-quality switches feature all-LED lighting. The anodized metal
work surface features rub-proof, etched markings that can’t rub off. Smooth, accurate rotary faders with
push-on/push-off channel switches make fast work of audio control. And RAQ features Axia’s famous
fully-automatic mix-minus for phone callers and codec sources, too. Show Profiles give instant recall of
up to 4 pre-defined console “snapshots”.
RAQ is ideal for standalone installation, but networks with larger Axia networks too. A RAQ control
surface and a QOR.16 integrated console engine constitute a complete RAQ system, but two RAQ
consoles, or one RAQ and one DESQ console, may be paired with a single QOR.16 for cost-effective
multi-console deployment.
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FEATURES
• 6 faders, each with instant access to any source. Assign any type of source to any channel with a
simple twist of the Options knob.
• Proven surface-and-core architecture separates control from mixing processes. No audio passes
directly through RAQ; all mixing and processing is performed in the QOR.16 Integrated Console Engine
– so studio “accidents” don’t turn into off-air events.
• Software upgrade for QOR.16 integrated console makes the RAQ AoIP console AES67-compliant.
• Two stereo mix buses and a Preview (cue) bus.
• Alpha-numeric OLED displays below each fader always show the current audio source with audio
confidence meter, and, when the Options knob is pressed, offer fast adjustment of fader gain trim, EQ,
pan and balance and other features without panel clutter or intimidating controls.
• Channel-input confidence meters assure talent of audio presence before taking sources to air.
• Built-in three-band per-source EQ.
• Each fader’s context-sensitive Soft key can be used to activate talkback, start delivery system events,
or perform other special functions.
• Every channel strip has a stereo Preview (“cue”) function, with a unique interlock system for fast cuing
of multiple sources.
• Reconfigurable CR monitor section with direct-selection of Program buses and reassignable buttons
that allow instant monitoring of external sources.
• Four custom Show Profile “snapshots” can be saved to instantly recall frequently-used console setups –
useful to quickly prepare for interview segments, music-intensive programming, call-in talk shows, etc.
• Automatic mix-minuses for phones, remote talent, etc.
• Bright OLED meter display provides responsive, readable VU or PPM metering styles. Switchable
display allows metering either Program bus.
• All functions can be accessed remotely for configuration, management and diagnostic purposes using
any standard Web browser.
• Network gateway enables loading networked sources while simultaneously exporting outputs back to
the network.
• Easy-to-deploy QOR.16 integrated console engine includes console CPU and power supply, DSP mixing
engine, custom Ethernet switch with 6 Livewire® ports and 2 Gigabit ports for studio networking, 8 analog
inputs and 4 analog outputs, 1 AES input and 1 AES output, 2 Mic inputs with switchable Phantom power,
and 4 GPIO ports for machine control. I/O can be expanded using Telos Alliance® xNodes.
• QOR.16’s integrated zero-configuration network switch is custom-designed for broadcasting — no
switch setup required. Supports Simple Networking, allowing up to 4 QOR engines to be daisy-chained
without the need for a separate core switch.
• Each QOR.16 can support two connected RAQ or DESQ consoles, or one of each.
• Fan-free, convection-cooled power supply for noiseless in-studio operation.
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IN DEPTH
A big console for small spaces.
Not every studio requires a full-size mixing console. Not every studio is full-size, itself! But you still want
the advantages of IP-Audio networking: the ability to send program audio to other studios, the ability to
consume audio from satellite downlinks, remote codecs and phone hybrids, or to trigger routing scene
changes from a user-mapped control panel. And you don’t want a toylike plastic pro-audio mixer — you
want a real broadcast console that fits into a rack or turret, or on a small desktop space. A console with a
small footprint, but big capabilities.
RAQ is a compact, special-purpose IP console from Axia. It may be compact in stature, but it’s big on
features and performance. RAQ has “big board” capabilities you won’t find in other consoles of this size
— automatic per-fader mix-minus, built-in EQ for voice and codec sources, and the ability to instantly
load new local or networked sources to any fader with just the turn of a knob. Which means RAQ easily
out-classes mixers with similar form factors — and even ones that take up much more room.
RAQ is a six-channel mixer over-engineered the Axia way, with super-duty rotary faders, aluminum
front-panel, high-resolution OLED displays for channel assignment and metering, heavy-duty switches
with LED lighting, and four Show Profile snapshot locations you can use to store and instantly recall
favorite console configurations. One touch, and presto! Talent’s favorite sources are loaded, monitor
source configured, and bus assignments made.
RAQ has two stereo mixing buses, plus a Preview (cue) bus, which makes it the perfect rack-mount
utility mixer, whether in the studio, in an OB van, or in a road case. It fits in just 4 RU of space, so you can
place it anywhere you need a full-featured, rack-mounted mixer: News booths, editors’ workstations,
voice-over booths, dubbing stations, even small remote studios or club installations.
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RAQ also features something else you won’t find on other compact consoles: a full-featured Monitor
section. Along with headphone and Preview volume controls, you’ll also find a selector that lets you
hear either Program 1, Program 2, or one of two External sources —helpful for monitoring off-air feeds,
a processed headphone chain, or another studio. And you can finally say goodbye to Dymo labels and
masking tape: each channel has an OLED display to show exactly what source is loaded.
Audio I/O, GPIO, console CPU, super-duty power supply, and even a network switch are all built into the
QOR.16. Just plug in your mics, CD players, codecs, profanity delays, whatever. There are 16 audio I/O
ports: two Mic inputs with switchable Phantom power, eight analog inputs and four analog outputs,
and one AES/EBU input and output. QOR.16 also has four GPIO logic ports for machine control of studio
peripherals, six 100BASE-T ports for Livewire devices, and two Gigabit ports with SFP for connection to
the outside world. For more I/O, just add Telos Alliance xNode interfaces. And you can daisy chain as
many as four QOR engines without the need for an external Ethernet switch, making installation even
more economical.
And here’s the kicker: one QOR.16 can power two RAQ mixers — or a RAQ and a DESQ (RAQ’s six-fader,
desktop-mount cousin)! Despite all these features, RAQ is so cost-effective, broadcasters are coming up
with creative, new uses for them. We figured folks would use them for news booths, dubbing stations
and guest performance mixers, but audio pros are also telling us they’d be ideal for broadcast remote
kits, mobile trucks, for shipboard broadcasting, or as personal mixers. What else could you use them for?
The possibilities are endless...
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SPECIFICATIONS
QOR.16 Connections
• Microphone Inputs: 2x balanced XLR-F, with selectable Phantom power
• Analog Inputs: 8x RJ-45, StudioHub+ standard.
• Analog Outputs: 4x RJ-45, StudioHub+ standard.
• AES/EBU Inputs: 1x RJ-45, StudioHub+ standard.
• AES/EBU Outputs: 1x RJ-45, StudioHub+ standard.
• GPIO: 4x DB-15
• Livewire:
• 4x 100BASE-T with PoE, RJ-45
• 2x 100BASE-T, RJ-45
• 2x 1000BASE-T, RJ-45
• 2x Gigabit, SFP (Small Form Pluggable)
• Console Frame Connections: 1x, 6-pin “latch and lock” style
• Accessory Connections: 1x, 6-pin “latch and lock” style
Microphone Preamplifiers
• Source Impedance: 150 Ohms
• Input Impedance: 4 k Ohms minimum, balanced
• Nominal Level Range: Adjustable, -75 dBu to -20 dBu
• Input Headroom: >20 dB above nominal input
• Output Level: +4 dBu, nominal
Analog Line Inputs
• Input Impedance: 20 k Ohms
• Nominal Level Range: Selectable, +4 dBu or -10dBv
• Input Headroom: 20 dB above nominal input
Analog Line Outputs
• Output Source Impedance: <50 Ohms balanced
• Output Load Impedance: 600 Ohms, minimum
• Nominal Output Level: +4 dBu
• Maximum Output Level: +24 dBu
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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, 20 kHz to 216kHz 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: 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: 125 dB
Equivalent Input Noise
• Microphone Preamp: -128 dBu, 150 Ohm source, reference -50 dBu input level
Total Harmonic Distortion + Noise
• Mic Pre Input to Analog Line Output: <0.005%, 1 kHz, -38 dBu input, +18 dBu output
• 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
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AXIA | RAQ
Crosstalk Isolation, Stereo Separation And CMRR
• Analog Line channel to channel isolation: 90 dB isolation minimum, 20 Hz to 20 kH
• Microphone channel to channel isolation: 80 dB isolation minimum, 20 Hz to 20 kHz
• Analog Line Stereo separation: 85 dB isolation minimum, 20Hz to 20 kHz
• Analog Line Input CMRR: >50 dB, 20 Hz to 20 kHz
• Microphone Input CMRR: >50 dB, 20 Hz to 20 kHz
Audio Processing
• Mic Equalizer (applicable to up to 6 faders)
• Frequency Bands: 20Hz to 320Hz, 125Hz to 2KHz, 1.25KHz to 20KHz.
• Cut/Boost range on each band: -25dB to +15dB.
• Q-factor: Automatic - bandwidth varies based on amount of cut or boost.
Power Supply AC Input, QOR.16 with RAQ Console
• Auto-sensing supply, 90VAC to 240VAC, 50 Hz to 60 Hz, IEC receptacle, internal fuse
• Power consumption: 100 Watts
Operating Temperatures
• -10 degrees C to +40 degrees C, <90% humidity, no condensation
Dimensions
• W 19.0 in (48 cm), H 3RU, 6.97 in (177.0 cm), D 2.54 in (64.50 cm)
Regulatory
North America: FCC and CE tested and compliant, power supply is UL approved.
Europe: Complies with the European Union Directive 2002/95/EC on the restriction of the use of certain
hazardous substances in electrical and electronic equipment (RoHS), as amended by Commission
Decisions 2005/618/EC, 2005/717/ EC, 2005/747/EC (RoHS Directive), and WEEE.
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AXIA | DESQ
DESQ
Compact Desktop IP Console
OVERVIEW
Six-fader, two-bus Axia DESQ console is a cost-effective, small-footprint console option perfect for
small production studios, remote vehicles, content ingest stations, etc. DESQ has two stereo mixing
buses and a Preview (cue) bus, six high-quality 100mm conductive-plastic faders for silky-smooth
operation and long life, razor-sharp OLED channel options displays, an OLED meter display with
switchable VU / PPM ballistics, and monitor / headphone controls for auditioning of Program buses or
two assignable External monitor source selections.
There’s also an OLED time-of-day clock + timer display, with auto / manual reset option. As with all
Axia consoles, aircraft-quality switches feature all-LED lighting; the anodized work surface has rubproof etched markings that can’t rub off. Additional features include automatic mix-minus for phone
callers and codec sources, EQ for voice sources, and Show Profile instant recall of up to four pre-defined
console “snapshots”.
DESQ requires no countertop cutout and takes only 16” square of desk space; it connects to the QOR.16
integrated console engine with a single power/control cable. DESQ is ideal for standalone installation,
but works with larger Axia networks too. A DESQ control surface and a QOR.16 integrated console
engine constitute a complete RAQ system, but two DESQ consoles, or one RAQ and one DESQ console,
may be paired with a single QOR.16 for cost-effective multi-console deployment.
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FEATURES
• 6 faders, each with instant access to any source. Assign any type of source to any channel with a
simple twist of the Options knob.
• Proven surface-and-core architecture separates control from mixing processes. No audio passes
directly through DESQ; all mixing and processing is performed in the QOR.16 Integrated Console
Engine – so studio “accidents” don’t turn into off-air events.
• Software upgrade for QOR.16 integrated console makes DESQ AoIP console AES67-compliant.
• Two stereo mix buses and a Preview (cue) bus.
• Alpha-numeric OLED displays below each fader always show the current audio source with audio
confidence meter, and, when the Options knob is pressed, offer fast adjustment of fader gain trim, EQ,
pan and balance and other features without panel clutter or intimidating controls.
• Channel-input confidence meters assure talent of audio presence before taking sources to air.
• Built-in three-band per-source EQ.
• Each fader’s context-sensitive Soft key can be used to activate talkback, start delivery system events,
or perform other special functions.
• Every channel strip has a stereo Preview (“cue”) function, with a unique interlock system for fast cuing
of multiple sources.
• Reconfigurable CR monitor section with direct-selection of Program buses and reassignable buttons
that allow instant monitoring of external sources.
• Four custom Show Profile “snapshots” can be saved to instantly recall frequently-used console setups –
useful to quickly prepare for interview segments, music-intensive programming, call-in talk shows, etc.
• Automatic mix-minuses for phones, remote talent, etc.
• Bright OLED meter display provides responsive, readable VU or PPM metering styles. Switchable
display allows metering either Program bus.
• All functions can be accessed remotely for configuration, management and diagnostic purposes using
any standard Web browser.
• Separate OLED clock/timer display features NTP-capable time-of-day clock and event timer that can
be manually or automatically reset via source activation.
• Network gateway enables loading networked sources while simultaneously exporting outputs back to
the network.
• Easy-to-deploy QOR.16 integrated console engine includes console CPU and power supply, DSP mixing
engine, custom Ethernet switch with 6 Livewire ports and 2 Gigabit ports for studio networking, 8 analog
inputs and 4 analog outputs, 1 AES input and 1 AES output, 2 Mic inputs with switchable Phantom power,
and 4 GPIO ports for machine control. I/O can be expanded using Telos Alliance® xNodes.
• QOR.16’s integrated zero-configuration network switch is custom-designed for broadcasting — no
switch setup required. Supports Simple Networking, allowing up to 4 QOR engines to be daisy-chained
without the need for a separate core switch.
• Each QOR.16 can support two connected RAQ or DESQ consoles, or one of each.
• Fan-free, convection-cooled power supply for noiseless in-studio operation.
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AXIA | DESQ
IN DEPTH
A big console for small spaces.
Not every studio requires a full-size mixing console. Not every studio is full-size, itself! But you still want
the advantages of IP-Audio networking: the ability to send program audio to other studios, the ability to
consume audio from satellite downlinks, remote codecs and phone hybrids, or to trigger routing scene
changes from a user-mapped control panel. And you don’t want a toylike plastic pro-audio mixer — you
want a real broadcast console that fits into a rack or turret, or on a small desktop space. A console with a
small footprint, but big capabilities.
DESQ is a compact, special-purpose IP console from Axia. It may be compact in stature, but it’s big on
features and performance. DESQ has “big board” capabilities you won’t find in other consoles of this size
— automatic per-fader mix-minus, built-in EQ for voice and codec sources, and the ability to instantly
load new local or networked sources to any fader with just the turn of a knob. Which means DESQ easily
out-classes mixers with similar form factors — and even ones that take up much more room.
DESQ is a six-fader console in a form-factor that lets it fit just about anywhere there’s a few inches of
spare space: DESQ is only 16 inches (39.9 cm) square. It’s built Axia-tough, with a machined-aluminum
work surface that takes the rough stuff jocks can dish out. Our familiar 100 mm. conductive-plastic
faders feel like silk under the fingertips, and you’ll also find the avionics-grade switches with LED lighting
that have become an Axia hallmark.
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AXIA | DESQ
Other features include OLED channel and meter displays, four-source monitor section with two external
locations that can be reassigned “on the fly”, and an OLED time-of-day clock and event timer. Like its
rackmount cousin, RAQ, DESQ also has four Show Profile console snapshot locations, and push-andturn Options knobs at the top of each fader that give instant access to fader source assignments, pan/
balance, and input gain trim.
Despite all these features, DESQ is so cost-effective, broadcasters are coming up with creative, new
uses for them. Its big features and small footprint make DESQ the perfect console for interview studios,
live performance spaces for on-air broadcast, news and feature production — whatever. Take it on road
trip remotes, or to sporting events where multiple mics are required. Or put it in mobile units or ENG kits.
Perfect for personal production studios, too.
Audio I/O, GPIO, console CPU, super-duty power supply, and even a network switch are all built into the
QOR.16. Just plug in your mics, CD players, codecs, profanity delays, whatever. There are 16 audio I/O
ports: two Mic inputs with switchable Phantom power, eight analog inputs and four analog outputs,
and one AES/EBU input and output. QOR.16 also has four GPIO logic ports for machine control of studio
peripherals, six 100Base-T ports for Livewire devices, and two Gigabit ports with SFP for connection
to the outside world. For more I/O, just add Telos Alliance xNode interfaces. And you can daisy chain as
many as four QOR engines without the need for an external Ethernet switch, making installation even
more economical. And here’s the kicker: one QOR.16 can power two DESQ mixers — or a DESQ and a
RAQ (DESQ’s six-fader, rackmount cousin)!
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AXIA | DESQ
SPECIFICATIONS
QOR.16 Connections
• Microphone Inputs: 2x balanced XLR-F, with selectable Phantom power
• Analog Inputs: 8x RJ-45, StudioHub+ standard.
• Analog Outputs: 4x RJ-45, StudioHub+ standard.
• AES/EBU Inputs: 1x RJ-45, StudioHub+ standard.
• AES/EBU Outputs: 1x RJ-45, StudioHub+ standard.
• GPIO: 4x DB-15
• Livewire:
• 4x 100Base-T with PoE, RJ-45
• 2x 100Base-T, RJ-45
• 2x 1000Base-T, RJ-45
• 2x Gigabit, SFP (Small Form Pluggable)
• Console Frame Connections: 1x, 6-pin “latch and lock” style
• Accessory Connections: 1x, 6-pin “latch and lock” style
Microphone Preamplifiers
• Source Impedance: 150 ohms
• Input Impedance: 4 k ohms minimum, balanced
• Nominal Level Range: Adjustable, -75 dBu to -20 dBu
• Input Headroom: >20 dB above nominal input
• Output Level: +4 dBu, nominal
Analog Line Inputs
• Input Impedance: 20 k Ohms
• Nominal Level Range: Selectable, +4 dBu or -10dBv
• Input Headroom: 20 dB above nominal input
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Analog Line Outputs
• 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, 20 kHz to 216kHz 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: 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: 125 dB
Equivalent Input Noise
• Microphone Preamp: -128 dBu, 150 ohm source, reference -50 dBu input level
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Total Harmonic Distortion + Noise
• Mic Pre Input to Analog Line Output: <0.005%, 1 kHz, -38 dBu input, +18 dBu output
• 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 kH
• Microphone channel to channel isolation: 80 dB isolation minimum, 20 Hz to 20 kHz
• Analog Line Stereo separation: 85 dB isolation minimum, 20Hz to 20 kHz
• Analog Line Input CMRR: >50 dB, 20 Hz to 20 kHz
• Microphone Input CMRR: >50 dB, 20 Hz to 20 kHz
Audio Processing
• Mic Equalizer (applicable to up to 6 faders)
• Frequency Bands: 20Hz to 320Hz, 125Hz to 2KHz, 1.25KHz to 20KHz.
• Cut/Boost range on each band: -25dB to +15dB.
• Q-factor: Automatic - bandwidth varies based on amount of cut or boost.
Power Supply AC Input, QOR.16 with DESQ Console
• Auto-sensing supply, 90VAC to 240VAC, 50 Hz to 60 Hz, IEC receptacle, internal fuse
• Power consumption: 100 Watts
Operating Temperatures
• -10 degrees C to +40 degrees C, <90% humidity, no condensation
Dimensions
• W 15.31in (38.9 cm), H 15.30 in (38.87 cm), D 2.79 in (7.1 cm)
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AXIA | STUDIOENGINE
Axia® StudioEngine
Bulletproof Power for 24/7 Operation.
OVERVIEW
The networked Axia StudioEngine provides bulletproof mixing console signal processing for Element®
and Fusion™ mixing consoles. Each StudioEngine is equipped with multiple simultaneous inputs, outputs,
mix-minus feeds, monitor signals, etc. and can provide EQ for multiple channels, voice dynamics, studio
headphone processing and multiple VMix (Virtual Mixer) channels.
The StudioEngine is fanless for cool, silent in-studio deployment, and is equipped with Gigabit Ethernet
ports for network connection, and dual-redundant, field-replaceable internal power supplies with
automatic switching for complete peace of mind.
In addition to providing mixing for physical console surfaces, StudioEngine can be used in conjunction
with Axia SoftSurface Virtual Console software to create a powerful “virtual console” of up to 48 faders
that’s controlled with a standard Windows-based laptop — perfect for places where space constraints
do not permit a physical mixing device.
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FEATURES
• Fanless design with heavy machined heat-sinks is completely silent in-studio.
• Front-panel OLED display delivers monitors power status, operating temperature, and a fast manual
setup option.
• Telecom grade dual-redundant power supplies are designed for maximum uptime under harsh
conditions. Automatic, seamless switching.
• Field-replaceable, internally fused fault-protected power modules change out in under 1 minute.
• Separate Gigabit LAN and WAN connections gives maximum network security while allowing
administrative remote access.
• Fusion 3.1 software update adds AES67 support.
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IN DEPTH
Mixing Power, a la carte.
Although Axia mixing consoles resemble traditional broadcast consoles, no audio is actually mixed by or
even passes through their faders. Instead, think of Axia consoles as a “remote control” for Axia DSPbased mixing engines.
The rugged StudioEngine is a standalone mixing engine for use with Axia Element and Fusion control
surfaces; it has no audio I/O of its own, instead allowing you maximum flexibility in designing your
audio network with a la carte I/O using Telos Alliance® xNode IP-Audio interfaces. In this way, you can
construct bespoke systems to suit your specific needs.
The StudioEngine itself is an extremely powerful mixing device, based on a blazingly-fast Intel processor
that can out-perform even the largest dedicated-DSP embedded designs. The StudioEngine accesses
audio streams, modifies them, and then presents the resulting streams back to the network as program
output (or monitor output, or mix-minus output, et cetera). This approach is ideally suited to a networkbased audio architecture since all input and output streams are routed through a Gigabit Ethernet port.
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AXIA | STUDIOENGINE
To deliver the reliability and ultra-low latency required, we equipped the StudioEngine with a fast, robust
version of the Linux real-time operating system. Then we optimized our engine processing program so
that total input to output latency is just a few hundred microseconds.
In fact, each StudioEngine has so much CPU power, it can outperform the very largest digital or routerbased consoles, with multiple simultaneous inputs, outputs, mix-minus feeds, monitor signals, etc.
for consoles as large as 40 faders. It can even provide EQ for and voice dynamics for multiple audio
channels, as well as multiple VMix (virtual mixer) channels that allow combination of multiple audio
channels on “virtual faders” that can then be mapped to a single physical fader. One StudioEngine
supplies mixing power for even the largest Fusion or Element console.
Additionally, StudioEngine can be paired with Axia SoftSurface software to create a “virtual console”
controlled by any computer with the Windows operating system — an ideal way of putting big mixing
power into very small spaces.
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SPECIFICATIONS
Power Supply AC Input
• Auto-sensing supply, 90VAC to 240VAC, 50 Hz to 60 Hz, IEC receptacle, internal fuse
• Power consumption: 100 Watts
Operating Temperatures
• -10 degrees C to +40 degrees C, <90% humidity, no condensation
Dimensions (HxWxD) and Weight
• 3.5 x 19 x 15 inches, 15 pounds
Network Interface
• 2x 1000BASE-T ports, standard RJ-45 connectors.
Audio Processing
Equalizer
• Frequency Bands: 20Hz to 320Hz, 125Hz to 2KHz, 1.25KHz to 20KHz.
• Cut/Boost range on each band: -25dB to +15dB.
• Q-factor: Automatic - bandwidth varies based on amount of cut or boost.
Compressor
• Threshold: -30dB to 0dB Ratio: 1:1 to 16:1
• Post-processor Trim Level: Adjustable from -20dB to +20dB
Expander/Noise Gate
• Threshold: -50dB to 0dB Ratio: -30dB to 0dB
De-esser
• Threshold: -20dB to 0dB Ratio: 1:1 to 8:1
Regulatory
North America: FCC and CE tested and compliant, power supply is UL approved.
Europe: Complies with the European Union Directive 2002/95/EC on the restriction of the use of certain
hazardous substances in electrical and electronic equipment (RoHS), as amended by Commission
Decisions 2005/618/EC, 2005/717/ EC, 2005/747/EC (RoHS Directive), and WEEE.
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AXIA | POWERSTATION
Axia® PowerStation®
Integrated Console Engine
OVERVIEW
PowerStation is an all-in-one studio solution that combines audio I/O, a console power supply, mixing
engine and built-for-broadcast network switch into one easy-to-deploy package. Each PowerStation
Main provides 4 Analog inputs and 6 Analog outputs, 2 AES/EBU inputs and 2 AES/EBU outputs,
2 Microphone inputs with selectable Phantom power, 4 GPIO machine-control logic ports, each
with 5 inputs and 5 outputs, an integrated network switch with 14 100BASE-T Ethernet ports and
2 1000BASE-T (Gigabit) ports with SFP, a heavy-duty Telecom-grade power supply with fanless
convection cooling, and an industrial-grade CPU designed for harsh-environment reliability.
Use PowerStation Main with an Element® or Fusion™ mixing console as a standalone studio solution, or
connect to other Axia equipment as part of a larger IP-Audio network. Simple Networking allows daisychain connection of up to 4 PowerStation-based studios without the use of an external network switch.
Connecting a PowerStation Aux adds auto-switching redundant backup power and doubles audio I/O
and GPIO capacity. I/O can also be easily expanded using Telos Alliance® xNodes.
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AXIA | POWERSTATION
FEATURES
• Fanless design with heavy machined heat-sinks is completely silent in-studio.
• Front-panel status display monitors power and network status.
• Telecom grade power supplies are designed for maximum uptime under harsh conditions.
• Add a PowerStation Aux to PowerStation Main for dual-redundant power supply with automatic,
seamless switching.
• Add redundant power to PowerStation Main without adding additional IO with Axia Console
Power Supply.
• Built-in, zero-configuration network switch with Gigabit and SFP for long-distance fiber connection.
• Large variety of built-in audio I/O boasts studio-grade audio performance specs.
• Add more I/O with PowerStation Aux, or a la carte using Telos Alliance xNodes.
• Fusion 3.1 software update adds AES67 support.
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AXIA | POWERSTATION
IN DEPTH
There’s no such thing as too much uptime.
If you set out to build a console engine designed to power your studio 24 hours a day, 7 days a week,
52 weeks a year, you probably wouldn’t skimp. You’d equip it with the most bulletproof, telecom-grade
power supply you could find. You’d give it a redundant-power option, for even more peace of mind. You’d
make it convection-cooled — no noisy cooling fans to assault your quiet studio. You’d give it plenty of
I/O — analog, digital, Mic-level and GPIO logic. And then, the pièce de résistance: you’d equip it with a
zero-configuration, built-for-broadcast Ethernet switch.
That’s what we did when we designed PowerStation, the muscle behind our industry-leading Fusion and
Element mixing consoles. PowerStation is over-engineered to Axia standards, every part chosen for its
ability to give constant, uninterrupted service. PowerStation combines four separate devices – a DSP
mixing engine, a console CPU and power supply, audio I/O, GPIO and a custom, Axia-designed Ethernet
switch – into a self-contained console engine that’s engineered to ensure years of reliable, troublefree service.
There are no compromises: PowerStation uses only best-of-the-best components, like studio-grade mic
preamps and 24-bit, 256x oversampling A/D converters, a rigid, EM-tight chassis, an ultra-reliable DSP
platform (not a common PC motherboard) and a hardened power supply designed for unfailing service,
even in the harshest environments.
PowerStation Main is where you start. Inside is a bulletproof mixing engine capable of powering consoles
of up to 40 faders. There’s a massive fanless, convection-cooled power supply. There are two Mic inputs,
four Analog inputs and six outputs, two AES/EBU inputs and two outputs, and four GPIO ports, each
with five opto-isolated inputs and five opto-isolated outputs. There are 14 100BASE-T Ethernet ports
with Livewire® for single-cable connection of Telos® phone systems, Omnia® audio processors and
other Axia equipment, as well as gear from our huge list of Livewire partners. Two Gigabit ports with SFP
enable connection to other studios via copper or fiber. Just connect it to your Element console (it only
takes a single cable), plug in your audio devices, and perform some fast web-based configuration. Add
power and you’re on the air. It’s that simple!
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AXIA | POWERSTATION
To beef up your PowerStation studio even further, there’s PowerStation Aux. Connect it to the
PowerStation main to instantly double mic, analog, AES and GPIO ports, and add a redundant backup
power supply with auto-switchover. Most redundant supplies protect only the console, but with
PowerStation, the mixing engine, audio I/O and network switch are protected as well. You can also
add redundant power to PowerStation Main without additional IO with Axia Console Power Supply,
which offers a single-cable connection to PowerStation Main, providing backup power with automatic
switching. (Auto-sensing power supply, 90VAC to 240VAC, 50 Hz to 60 Hz. 250 Watts, 2RU.)
Best of all, there’s that zero-configuration Ethernet switch that’s built specifically to handle IP-Audio.
No settings to tweak, no configuration code to upload – just plug it in and go. There are even two Gigabit
ports with SFP, to connect to other studios via fiber or copper. You can even daisy-chain up to four
PowerStation studios directly, for a self-contained network that doesn’t require an external Ethernet
switch. No other console company makes AoIP this easy.
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SPECIFICATIONS
Microphone Preamplifiers
• Source Impedance: 150 Ohms
• Input Impedance: 4 k Ohms minimum, balanced
• Nominal Level Range: Adjustable, -75 dBu to -20 dBu
• Input Headroom: >20 dB above nominal input
• Output Level: +4 dBu, nominal
Analog Line Inputs
• Input Impedance: >40 k Ohms, balanced
• Nominal Level Range: Selectable, +4 dBu or -10dBv
• Input Headroom: 20 dB above nominal input
Analog Line Outputs
• 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 Ohms, 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
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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
Equivalent Input Noise
• Microphone Preamp: -128 dBu, 150 ohm source, reference -50 dBu input level
Total Harmonic Distortion + Noise
• Mic Pre Input to Analog Line Output: <0.005%, 1 kHz, -38 dBu input, +18 dBu output
• 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 kH
• Microphone channel to channel isolation: 80 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
• Microphone Input CMRR: >55 dB, 20 Hz to 20 kHz
Audio Processing
Equalizer
• Frequency Bands: 20Hz to 320Hz, 125Hz to 2KHz, 1.25KHz to 20KHz.
• Cut/Boost range on each band: -25dB to +15dB.
• Q-factor: Automatic - bandwidth varies based on amount of cut or boost.
Compressor
• Threshold: -30dB to 0dB Ratio: 1:1 to 16:1
• Post-processor Trim Level: Adjustable from -20dB to +20dB
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AXIA | POWERSTATION
Expander/Noise Gate
• Threshold: -50dB to 0dB Ratio: -30dB to 0dB
De-esser
• Threshold: -20dB to 0dB Ratio: 1:1 to 8:1
Power Supply AC Input, PowerStation Aux & Main
• Auto-sensing supply, 90VAC to 240VAC, 50 Hz to 60 Hz, IEC receptacle, internal fuse
• Power consumption: 500 Watts
Axia Console Power Supply
• Add redundant power to PowerStation main without additional IO.
• Single-cable connection to PowerStation main provides backup power with automatic switching.
• Auto-sensing power supply, 90VAC to 240VAC, 50 Hz to 60 Hz.
• Power consumption: 250 Watts. Operating Temperatures
• -10 degrees C to +40 degrees C, <90% humidity, no condensation
Dimensions (HxWxD) and Weight
• PowerStation Main/Aux: 7 x 19 x 15.5 inches (behind rail)
• Front panel extends 2.25 inches in front of rack rail
• PowerStation Main: 45 pounds
• PowerStation Aux: 40 pounds
Regulatory
North America: FCC and CE tested and compliant, power supply is UL approved.
Europe: Complies with the European Union Directive 2002/95/EC on the restriction of the use of certain
hazardous substances in electrical and electronic equipment (RoHS), as amended by Commission
Decisions 2005/618/EC, 2005/717/ EC, 2005/747/EC (RoHS Directive), and WEEE.
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AXIA | QOR.32
Axia® QOR.32
Integrated Console Engine
OVERVIEW
QOR.32 is an Axia integrated console engine for iQ mixing consoles that combines audio I/O, a console
power supply, mixing engine and built-for-broadcast network switch into one easy-to-deploy package.
Each QOR.32 provides 16 Analog inputs and 8 Analog outputs, 2 AES/EBU inputs and 2 AES/EBU
outputs, 4 Microphone inputs with selectable Phantom power, 8 GPIO machine-control logic ports, each
with 5 inputs and 5 outputs, an integrated network switch with 6 Livewire® 100BASE-T Ethernet ports
and 2 1000BASE-T (Gigabit) ports with SFP, a heavy-duty Telecom-grade power supply with fanless
convection cooling, and an industrial-grade CPU designed for harsh-environment reliability.
Use QOR.32 with an Axia iQ mixing console as a standalone studio solution, or connect to other Axia
equipment as part of a larger IP-Audio network. Simple Networking allows daisy-chain connection of up
to 4 QOR-based studios without the use of an external network switch. Connecting a QOR Backup adds
auto-switching redundant backup power. I/O can easily be expanded using Axia Audio Nodes.
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AXIA | QOR.32
FEATURES
• Fanless design with heavy machined heat-sinks is completely silent in-studio.
• Front-panel LED display monitors power and network status.
• Telecom grade power supplies are designed for maximum uptime under harsh conditions.
• Add an Axia Console Power Supply Backup to QOR.32 for dual-redundant power supply with
automatic, seamless switching.
• Built-in, zero-configuration network switch with Gigabit and SFP for long-distance fiber connection.
• Large variety of built-in audio I/O boasts studio-grade audio performance specs.
• Add more I/O a la carte using Telos Alliance® xNodes.
• Software upgrade adds AES67 support, allowing the QOR.32 integrated console engine to receive and
transmit AES67 streams via Livewire+™ AES67.
IN DEPTH
QOR.32 Integrated Console Engine
The QOR.32 integrated console engine is a DSP-based mixing engine with onboard I/O, GPIO, console
power supply and custom-built, configuration-free Ethernet switch. You’ll find plenty of I/O, including
mic inputs with selectable Phantom power, analog and AES/EBU inputs and outputs, plenty of GPIO
machine-control logic ports, and that powerful integrated Ethernet switch with Livewire ports to add
local sources, and Gigabit ports for networking with the rest of your plant. That’s a lot of I/O, but if you
need more you can instantly add it just by plugging in Telos Alliance xNode audio interfaces. And QOR.32
is convection-cooled for utterly silent, fan-free operation.
Let’s take a look around back, shall we? You’ll find everything you need for an average, medium-sized
studio: 4 mic inputs with selectable Phantom power, 16 stereo analog inputs and 8 stereo analog
outputs, 2 AES/EBU inputs and 2 AES/EBU outputs, 8 GPIO machine-control logic ports (each with 5
opto-isolated inputs and 5 outputs).
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AXIA | QOR.32
There’s Livewire I/O as well: the QOR.32 has an integrated Ethernet switch with 6 Livewire 100BASE-T
ports. 4 of those ports have PoE (Power over Ethernet) that you can use to connect and power
networked devices compatible with the IEEE 802.1af PoE standard (like our xNode audio interfaces,
or Telos VSet phones). You’ll also find 2 1000BASE-T Gigabit ports (RJ-45 & SFP) that you can use to
connect to other studios. 4 CANBus ports provide for connection of up to 3 Axia iQ console frames,
allowing construction of consoles up to 24 faders in size.
By the way, that zero-configuration Ethernet switch is built specifically to handle IP-Audio. No settings
to tweak, no configuration code to upload – just plug it in and go. The built-in configurable network
gateway allows loading sources from other studios, while simultaneously exporting audio streams for
use elsewhere; the gateway can be configured for 12-in, 4-out or 8-in, 8-out modes. You can even
daisy-chain up to four QOR-based studios directly, for a self-contained network that doesn’t require an
external Ethernet switch. No other console company makes AoIP this easy.
For installations that require redundant backup power, there’s the Axia Console Power Supply. Connect it
to the QOR.32 and you’ve added a redundant backup power supply with auto-switchover. Single-cable
connection to QOR.32 console engine provides backup power with automatic switching. (Auto-sensing
power supply, 90VAC to 240VAC, 50 Hz to 60 Hz. 250 Watts. Rackmount, 2RU.) Most redundant supplies
protect only the console, but with Axia’s integrated console engine concept, the mixing engine, local
audio I/O and network switch are protected as well.
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AXIA | QOR.32
SPECIFICATIONS
QOR.32 Connections
• Microphone Inputs: 4x balanced XLR-F, with selectable Phantom power
• Analog Inputs: 16x RJ-45, StudioHub+ standard.
• Analog Outputs: 8x RJ-45, StudioHub+ standard.
• AES/EBU Inputs: 2x RJ-45, StudioHub+ standard.
• AES/EBU Outputs: 2x RJ-45, StudioHub+ standard.
• GPIO: 8x DB-15
• Livewire:
• 4x 100BASE-T with PoE, RJ-45
• 2x 100BASE-T, RJ-45
• 2x 1000BASE-T, RJ-45
• 2x Gigabit, SFP (Small Form Pluggable)
• Console Frame Connections: 3x, 6-pin “latch and lock” style
• Accessory Connections: 1x, 6-pin “latch and lock” style
Microphone Preamplifiers
• Source Impedance: 150 ohms
• Input Impedance: 4 k ohms minimum, balanced
• Nominal Level Range: Adjustable, -75 dBu to -20 dBu
• Input Headroom: >20 dB above nominal input
• Output Level: +4 dBu, nominal
Analog Line Inputs
• Input Impedance: 20 k Ohms
• Nominal Level Range: Selectable, +4 dBu or -10dBv
• Input Headroom: 20 dB above nominal input
Analog Line Outputs
• Output Source Impedance: <50 ohms balanced
• Output Load Impedance: 600 ohms, minimum
• Nominal Output Level: +4 dBu
• Maximum Output Level: +24 dBu
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AXIA | QOR.32
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, 20 kHz to 216kHz 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: 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: 125 dB
Equivalent Input Noise
• Microphone Preamp: -128 dBu, 150 ohm source, reference -50 dBu input level
Total Harmonic Distortion + Noise
• Mic Pre Input to Analog Line Output: <0.005%, 1 kHz, -38 dBu input, +18 dBu output
• 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
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AXIA | QOR.32
Crosstalk Isolation, Stereo Separation And CMRR
• Analog Line channel to channel isolation: 90 dB isolation minimum, 20 Hz to 20 kH
• Microphone channel to channel isolation: 80 dB isolation minimum, 20 Hz to 20 kHz
• Analog Line Stereo separation: 85 dB isolation minimum, 20Hz to 20 kHz
• Analog Line Input CMRR: >50 dB, 20 Hz to 20 kHz
• Microphone Input CMRR: >50 dB, 20 Hz to 20 kHz
Audio Processing
• Mic Equalizer (applicable to up to 6 faders)
• Frequency Bands: 20Hz to 320Hz, 125Hz to 2KHz, 1.25KHz to 20KHz.
• Cut/Boost range on each band: -25dB to +15dB.
• Q-factor: Automatic - bandwidth varies based on amount of cut or boost.
Power Supply AC Input, QOR.32
• Auto-sensing supply, 90VAC to 240VAC, 50 Hz to 60 Hz, IEC receptacle, internal fuse
• Power consumption: 100 Watts
Axia Console Power Supply
• Auto-sensing supply, 90VAC to 240VAC, 50 Hz to 60 Hz, IEC receptacle, internal fuse
• Power consumption: 250 Watts
Operating Temperatures
• -10 degrees C to +40 degrees C, <90% humidity, no condensation
Regulatory
North America: FCC and CE tested and compliant, power supply is UL approved.
Europe: Complies with the European Union Directive 2002/95/EC on the restriction of the use of certain
hazardous substances in electrical and electronic equipment (RoHS), as amended by Commission
Decisions 2005/618/EC, 2005/717/ EC, 2005/747/EC (RoHS Directive), and WEEE.
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AXIA | QOR.16
Axia® QOR.16
Integrated Console Engine
OVERVIEW
QOR.16 is an Axia integrated console engine for Radius, DESQ and RAQ mixing consoles. QOR.16
combines audio I/O, a console power supply, mixing engine and built-for-broadcast network switch into
one easy-to-deploy package. Each QOR.16 provides 8 Analog inputs and 4 Analog outputs, 1 AES/EBU
input and 1 AES/EBU output, 2 Microphone inputs with selectable Phantom power, 4 GPIO machinecontrol logic ports, each with 5 inputs and 5 outputs, an integrated network switch with 6 Livewire®
100BASE-T Ethernet ports and 2 1000BASE-T (Gigabit) ports with SFP, a heavy-duty Telecom-grade
power supply with fanless convection cooling, and an industrial-grade CPU designed for harshenvironment reliability.
Use QOR.16 with a Radius, DESQ or RAQ mixing console as a standalone studio solution, or connect
to other Axia equipment as part of a larger IP-Audio network. Simple Networking allows daisy-chain
connection of up to 4 QOR-based studios without the use of an external network switch. I/O can easily
be expanded using Telos Alliance® xNodes.
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AXIA | QOR.16
FEATURES
• Fanless design with heavy machined heat-sinks is completely silent in-studio.
• Front-panel LED display monitors power and network status.
• Telecom grade power supply is designed for maximum uptime under harsh conditions.
• PoE (Power over Ethernet) capability can supply power for PoE-compliant studio devices.
• Built-in, zero-configuration network switch with Gigabit and SFP for long-distance fiber connection.
• Large variety of built-in audio I/O boasts studio-grade audio performance specs.
• Add more I/O a la carte using Telos Alliance xNodes.
• Software upgrade adds AES67 support, allowing the QOR.16 integrated console engine to receive and
transmit AES67 streams via Livewire+™ AES67.
IN DEPTH
QOR.16 Integrated Console Engine
The QOR.16 integrated console engine is a DSP-based mixing engine with onboard I/O, GPIO, console
power supply and custom-built, configuration-free Ethernet switch. It’s the smaller brother of our
QOR.32 integrated console engine, designed and built with the same high-grade components for
deployment with Radius, DESQ and RAQ consoles in smaller studios where large amounts of I/O are not
required.
QOR.16 comes with a wide variety of I/O, including mic inputs with selectable Phantom power, analog
and AES/EBU inputs and outputs, plenty of GPIO machine-control logic ports, and that powerful
integrated Ethernet switch with Livewire ports to add local sources, and Gigabit ports for networking
with the rest of your plant. If more you I/O is needed, you can instantly add it just by plugging in Telos
Alliance xNode audio interfaces. And QOR.16 is convection-cooled for utterly silent, fan-free operation.
QOR.16 has all the analog and digital inputs and outputs an average small studio requires: 2 mic
inputs with selectable Phantom power, 8 stereo analog inputs and 4 stereo analog outputs, 1 AES/
EBU input and 1 AES/EBU output, and 4 GPIO machine-control logic ports (each with 5 opto-isolated
inputs and 5 outputs).
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AXIA | QOR.16
Of course, Livewire connections are built in. The QOR.16 has an integrated Ethernet switch with 6
Livewire 100BASE-T ports. 4 of those ports have PoE (Power over Ethernet) that you can use to connect
and power networked devices compatible with the IEEE 802.1af PoE standard (like our xNode audio
interfaces, or Telos VSet phones). You’ll also find 2 1000BASE-T Gigabit ports (RJ-45 & SFP) that you can
use to connect to other studios.
By the way, that zero-configuration Ethernet switch is built specifically to handle IP-Audio. No settings
to tweak, no configuration code to upload – just plug it in and go. The built-in configurable network
gateway allows loading sources from other studios, while simultaneously exporting audio streams for
use elsewhere; the gateway can be configured for 12-in, 4-out or 8-in, 8-out modes. You can even
daisy-chain up to four QOR-based studios directly, for a self-contained network that doesn’t require an
external Ethernet switch. No other console company makes AoIP this easy.
And here’s a neat trick: if you’re building audio workstations, news bullpens or ingest facilities, where
small consoles like Axia DESQ or RAQ shine, a single QOR.16 can provide mixing power for two DESQ or
RAQ mixers — or one of each! Just another way choosing Axia helps stretch your equipment budget.
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AXIA | QOR.16
SPECIFICATIONS
QOR.16 Connections
• Microphone Inputs: 2x balanced XLR-F, with selectable Phantom power
• Analog Inputs: 8x RJ-45, StudioHub+ standard.
• Analog Outputs: 4x RJ-45, StudioHub+ standard.
• AES/EBU Inputs: 1x RJ-45, StudioHub+ standard.
• AES/EBU Outputs: 1x RJ-45, StudioHub+ standard.
• GPIO: 4x DB-15
• Livewire:
• 4x 100BASE-T with PoE, RJ-45
• 2x 100BASE-T, RJ-45
• 2x 1000BASE-T, RJ-45
• 2x Gigabit, SFP (Small Form Pluggable)
• Console Frame Connections: 1x, 6-pin “latch and lock” style
• Accessory Connections: 1x, 6-pin “latch and lock” style
Microphone Preamplifiers
• Source Impedance: 150 ohms
• Input Impedance: 4 k ohms minimum, balanced
• Nominal Level Range: Adjustable, -75 dBu to -20 dBu
• Input Headroom: >20 dB above nominal input
• Output Level: +4 dBu, nominal
Analog Line Inputs
• Input Impedance: 20 k Ohms
• Nominal Level Range: Selectable, +4 dBu or -10dBv
• Input Headroom: 20 dB above nominal input
Analog Line Outputs
• Output Source Impedance: <50 ohms balanced
• Output Load Impedance: 600 ohms, minimum
• Nominal Output Level: +4 dBu
• Maximum Output Level: +24 dBu
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AXIA | QOR.16
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, 20 kHz to 216kHz 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: 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: 125 dB
Equivalent Input Noise
• Microphone Preamp: -128 dBu, 150 ohm source, reference -50 dBu input level
Total Harmonic Distortion + Noise
• Mic Pre Input to Analog Line Output: <0.005%, 1 kHz, -38 dBu input, +18 dBu output
• 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
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AXIA | QOR.16
Crosstalk Isolation, Stereo Separation And CMRR
• Analog Line channel to channel isolation: 90 dB isolation minimum, 20 Hz to 20 kH
• Microphone channel to channel isolation: 80 dB isolation minimum, 20 Hz to 20 kHz
• Analog Line Stereo separation: 85 dB isolation minimum, 20Hz to 20 kHz
• Analog Line Input CMRR: >50 dB, 20 Hz to 20 kHz
• Microphone Input CMRR: >50 dB, 20 Hz to 20 kHz
Audio Processing
• Mic Equalizer (applicable to up to 6 faders)
• Frequency Bands: 20Hz to 320Hz, 125Hz to 2KHz, 1.25KHz to 20KHz.
• Cut/Boost range on each band: -25dB to +15dB.
• Q-factor: Automatic - bandwidth varies based on amount of cut or boost.
Power Supply AC Input, QOR.16 With Radius Console
• Auto-sensing supply, 90VAC to 240VAC, 50 Hz to 60 Hz, IEC receptacle, internal fuse
• Power consumption: 100 Watts
Operating Temperatures
• -10 degrees C to +40 degrees C, <90% humidity, no condensation
Regulatory
North America: FCC and CE tested and compliant, power supply is UL approved.
Europe: Complies with the European Union Directive 2002/95/EC on the restriction of the use of certain
hazardous substances in electrical and electronic equipment (RoHS), as amended by Commission
Decisions 2005/618/EC, 2005/717/ EC, 2005/747/EC (RoHS Directive), and WEEE.
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TELOS ALLIANCE | XNODE
Telos Alliance® xNode™
IP-Audio Interfaces
The most advanced AoIP interfaces on the planet.
OVERVIEW
The xNode lightweight, half-rack, high-performance IP-Audio interface from Telos Alliance is loaded with
advanced features and capabilities. One-button configuration takes a new xNode from out-of-the-box to
on-the-air in under one minute. They’re fanless, which means they’re noiseless too. Versatile mounting
options let you deploy two xNodes in just 1RU of rack space, or on ceilings, walls, and under counters
with an available wall-mount kit. xNodes have studio-grade audio performance specs. Redundant
power options (using AC mains and Power-over-Ethernet) and dual-redundant network interfaces are
included, both with automatic switching. And xNodes are fully AES67-compliant, so they work with all
AES67 audio gear—now, and in the future. In fact, they are the first and only AoIP I/O devices that are
Livewire+™ AES67, RAVENNA, and AES67 compliant. Every xNode not only supports RAVENNA audio
stream interoperability, but also enables advertising/discovery of those streams natively, above and
beyond AES67.
xNodes are available in Analog, AES/EBU, Microphone-level, Mixed-Signal, and GPIO versions to handle
virtually any signal encountered in today’s broadcast studio.
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TELOS ALLIANCE | XNODE
FEATURES
• Fanless design with cast-aluminum heat-sinks is completely silent in-studio. Front-panel heat sinks
are cooled by ambient air, not “rack air,” eliminating overheating worries.
• World’s only fully AES67-compliant AoIP interface; xNodes are “universal translators” that support
a huge installed base of Livewire+™ AES67 hardware as well as audio streams from other AES67compliant devices.
• First and only AoIP I/O device that is Livewire+™ AES67, RAVENNA, and AES67 compliant. Every
xNode not only supports RAVENNA audio stream interoperability, but also enables advertising/
discovery of those streams natively, above and beyond AES67.
• High-resolution front-panel multi-function OLED display meters inputs and outputs or GPIO status,
gives software and other status information.
• Power-efficient: xNodes use just 14 Watts each.
• Exclusive redundant power plan uses AC and Power over Ethernet (IEEE 802.3af) supplied by
compliant Ethernet switches. Multi-color front-panel LED glows green when AC mains power is used,
red when PoE is used, and orange when both AC and PoE are connected.
• Exclusive redundant network connection: Dual NICs allow you to connect xNode to separate network
branches for full audio pathway redundancy. Automatic failover activates backup connection should
the primary be interrupted.
• Built-in Syslog server with configurable event filter and SNMP (Simple Network Management Protocol)
support help you stay fully informed, should an xNode’s power or connection status change.
• Synchronize your AES master clock to a designated xNode AES/EBU input to keep all of your AES
streams synchronized to the house clock.
• xNodes use premium components, including rugged cast aluminum faceplates and heat sinks,
high-resolution OLED displays, bulletproof power supplies designed for high-availability telecom
applications, studio-quality SRCs with recording-studio specs.
• I/O connections via industry-standard RJ-45 audio connectors or high-density DB-25 connections,
both available prefabricated and ready to attach in seconds.
• Versatile mounting options: Use freestanding, rack singly or side-by-side in 1 RU, or wall-mount using
an optional surface-mount kit.
• Analog xNode inputs can be configured to supply four stereo audio channels, eight true mono
channels, or 5.1 surround + stereo downmix. Outputs support the same variety of selections, easily
selectable in software via the built-in web interface.
• On the Analog, AES/EBU, Mixed Signal, and Microphone xNodes, a fully configurable mixing matrix
allows for mixing of both physical and network inputs, stream conversion, and a multitude of other
unique solutions.
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TELOS ALLIANCE | XNODE
IN DEPTH
The AoIP Interface that’s twice as powerful.
(But only half the size.)
One day, all audio equipment will be networked. Until then, there are xNodes, the world’s first selfconfiguring, fully AES67-compliant AoIP interfaces.
xNodes give you an easy way to add non-networked audio devices to your studio network. They pack a
lot of I/O into a very small space. And xNodes are so simple to set up, they nearly configure themselves.
All xNodes feature a high-resolution OLED front panel display and two “soft” buttons to provide status
information and assist with initial setup, and a multi-color LED that gives at-a-glance information about
the xNode’s power configuration. To ensure ultra-reliable network operations and extremely low delay,
xNodes run Linux on an embedded processor, and a built-in web server in each node gives you remote
configuration and control—in an intuitive, easy-to-understand manner—using any standard web browser.
xNodes are loaded with features designed to ensure the uptime of your network. Dual Ethernet ports
can provide redundant connections to separate network segments. Redundant power capability with
automatic switchover enables xNodes to run on house mains or PoE (Power over Ethernet), letting the
network switch itself supply power, and enabling easy single-cable setup in places where AC power
isn’t practical. Built-in Syslog servers with a configurable event filter and SNMP (Simple Network
Management Protocol) support let you stay fully informed, should an xNode’s power or connection
status change.
The xNode Matrix Mixer feature is one of the most flexible and capable virtual mixers available. It lets
users mix physical inputs (like mics and playback devices) with digital network sources (like stream
inputs) to a single output. With the xNode Matrix Mixer feature, broadcasters can bypass the studio
console during automated dayparts and send on-air mixes straight to the transmitter thus simplifying
audio workflows. This one-of-a-kind solution offers the power and flexibility of a big studio mixer
switching system in a compact ½ RU device!
xNodes are convenient, too. For example, a Microphone xNode placed in a studio can take audio from
microphones and also provide outputs to associated studio monitors and headphones. An xNode in the
rack room can collect audio from network feeds, codecs and other shared sources for system-wide use
while providing handy outputs for audio processors and other terminal-room gear.
xNodes provide audio quality superior to any other AoIP interface. Not only are they capable of operating
at a network sampling rate of 48kHz, they also employ high-resolution 32-bit floating-point SRC chips.
xNodes produce a “sweeter,” more natural audio quality—clients routinely tell us of noticeable sonic
improvements after installation.
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TELOS ALLIANCE | XNODE
xNodes are versatile and cost-efficient. Since they’re half the size of other AoIP interfaces, they cost
less. And you can mix-and-match I/O as needed: Choose between analog, AES/EBU, or Mic-level inputs,
without paying for ports you won’t use. High-density GPIO xNodes let you easily provide logic and
control for your audio source devices.
xNodes are easy to deploy, too. When you connect an xNode to your network, it automatically prompts
you to give it an ID via the front-panel controls. Then, it derives a unique static IP address, and even gives
names to its sources and outputs (which you can edit later, from the comfort of your computer). All you
have to do is connect devices to the inputs, and it advertises that its audio sources are available for use,
allowing any users access to them.
xNodes are also fanless, so you can tuck one anywhere you need I/O without worrying about cooling
fans or heat—they consume only 14 Watts of power! Two xNodes fit side-by-side in a single rack space
using the included rack-mount kit. Or, mount them to walls, ceilings, or under countertops, with an
optional surface-mount kit.
Five different xNodes provide analog and AES ins and outs, microphone inputs and GPIO logic ports,
wherever you need them. No need for “home runs” to a central rack—one CAT-5 cable connection is all
an xNode needs to interface multiple channels of bi-directional audio to your network.
Microphone xNode
The Microphone xNode has four professional-grade microphone preamps with selectable Phantom
power and software-adjustable gain. There are also four balanced analog line outputs to conveniently
deliver headphone and studio monitor feeds back to your talent. Inputs and outputs are presented both
on easy-to-install RJ-45s and high-density DB-25s, both of which connect to easily available 3rd-party
breakout cables, to suit your connection preference.
Analog xNode
The Analog xNode has 8 mono or 4 stereo balanced line-level inputs and 8 mono or 4 stereo balanced
line-level outputs, on RJ-45 and DB-25 connectors. It can also accommodate 5.1 Surround inputs and
outputs, each with an associated discrete Stereo mix. Each input is switchable to accommodate either
consumer-level -10dBv or professional level +4dBu sources. The short-circuit protected outputs can
deliver up to +24dBu before clipping. Telos Alliance uses only studio-grade A/D/A converters and low-noise
components, so that each Analog node provides superior audio performance for high-end studio use.
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TELOS ALLIANCE | XNODE
AES/EBU xNode
Our AES/EBU xNode has 4 AES/EBU inputs and 4 AES/EBU outputs. Left and right input signals may be
split and routed independently as mono signals. Stunning performance specs include 48 kHz sampling
rate, 126dB of dynamic range, and <0.0003% THD. Sample rate conversion is available on all inputs; the
unit can also be synchronized to a house clock to provide sync to your entire Axia network.
Mixed-Signal xNode
The Mixed-Signal xNode is your utility player; perfect for places that require a mix of different audio
I/O types. It provides 1 selectable Mic/Line analog input, 2 dedicated analog line inputs, 3 analog line
outputs, 1 digital AES3 input and 1 AES3 output, and 2 GPIO ports – truly a “jack of all trades.”
GPIO xNode
GPIO xNode provides 6 general-purpose logic ports for machine control of studio peripherals – audio
devices, loudspeaker muting relays, signal lamps, etc. – each with 5 opto-isolated inputs and 5 outputs.
A logic port can be associated with any audio input or output and routes control data transparently along
with the audio.
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TELOS ALLIANCE | XNODE
SPECIFICATIONS
Microphone Preamplifiers
• Source Impedance: 150 Ohms
• Input Impedance: 4k Ohms minimum, balanced
• Nominal Level Range: Adjustable, -75 dBu to -20 dBu
• Input Headroom: >20 dB above nominal input
• Phantom power: +48VDC, switchable
Analog Line Inputs
• Input Impedance: >40k Ohms, balanced
• Nominal Input Range: Selectable, +4 dBu or -10dBv
• Input Headroom: 20 dB above nominal input
Analog Line Outputs
• 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
• Signal Format: AES3 (AES/EBU)
• AES3 Input Compliance: 24-bit with sample rate conversion
• AES3 Output Compliance: 24-bit
• Digital Reference: Internal (network timebase) or external reference 48 kHz, +/- 2 ppm
• Internal Sampling Rate: 48 kHz
• Input Sample Rate: 32 kHz to 192kHz
• Output Sample Rate: 44.1 kHz or 48kHz
• A/D Conversions: 24-bit, Delta-Sigma, 256x oversampling
• D/A Conversions: 24-bit, Delta-Sigma, 256x oversampling
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TELOS ALLIANCE | XNODE
Frequency Response
• Any Input to Any Output: +/- 0.5 dB, 20 Hz to 20 kHz
Latency
• Analog Input to Analog Output, 2.75ms including network, converters, and mixing process
• Digital Input to Digital Output, 1.75ms including network mixing engine (ASRC off)
Dynamic Range
• Analog Inputs to Analog Outputs 108dB referenced to 0dBFs, 111dB A-weighted
• Analog Inputs to Digital Outputs 110dB referenced to 0dBFs, 113dB A-weighted
• Digital Inputs to Analog Outputs 112dB referenced to 0dBFs, 115dB A-weighted
• Digital Inputs to Digital Outputs 126dB
Equivalent Input Noise
• Microphone Preamp: -128 dBu, 150 Ohm source, reference -50 dBu input level
Total Harmonic Distortion + Noise
• Mic Pre Input to Analog Output: < 0.005%, 1 kHz, -36dBu input, +18dBu output
• Analog Input to Analog Output: < 0.005%, 1 kHz, +18dBu input, +18dBu output
• Analog Input to Digital Output: < 0.004%, 1 kHz, +18dBu input, -6dBFs output
• Digital Input to Analog Output: < 0.004%, 1 kHz, -6dBFs input, +18dBu output
• Digital Input to Digital Output: < 0.0003%, 1 kHz, -20dBFs
Crosstalk Isolation, Stereo Separation and CMRR
• Analog Line Channel to Channel Isolation: 90dB minimum, 20Hz to 20kHz
• Analog Line Stereo Separation: 85dB minimum, 20Hz to 20kHz
• Analog Line Input CMRR: 80dB minimum, 20Hz to 20kHz
• Microphone Input CMRR: >60 dB, 20 Hz to 20 kHz
Power Supply AC Input
• Auto-Ranging Supply, 95VAC to 240VAC, 50Hz to 60Hz, IEC Receptacle, Internal Fuse
• Power Consumption: 14 Watts
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TELOS ALLIANCE | XNODE
Operating Temperatures
• 0 degree C to +40 degree C, <90% humidity, no condensation
Dimensions
• 8.5” (22 cm) wide; two may be mounted side-by-side in a standard 1RU rack space; 1.72” (4.4 cm)
height, 11.75” (30 cm) depth
Regulatory
North America: FCC and CE tested and compliant, power supply is UL approved.
Europe: Complies with the European Union Directive 2002/95/EC on the restriction of the use of certain
hazardous substances in electrical and electronic equipment (RoHS), as amended by Commission
Decisions 2005/618/EC, 2005/717/ EC, 2005/747/EC (RoHS Directive), and WEEE.
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TELOS ALLIANCE | XSWITCH
Telos Alliance® xSwitch
The Network Switch Built for IP-Audio
OVERVIEW
xSwitch is the world’s only zero-configuration Ethernet switch optimized for Livewire® IP-Audio
applications. Fast setup requires only IP address assignment via front-panel OLED display or Axia®
iProbe software. Features 8 10/100MBit Ethernet ports — 4 with Power-over-Ethernet to power Telos
Alliance xNodes, Telos® VSet phones, and other networked devices compatible with the IEEE 802.1af
PoE standard. Two Gigabit ports are provided for trunking, both with RJ-45 (copper) and SFP (fiber)
connections; supports redundant copper/SFP Gigabit connections with auto-switching. Supports 2,000
Multicast groups and 2,000 ARP table entries (8x more than other small-form Ethernet switches).
Web-based management interface uses built-in HTTP server. 9.5” x 11” half-rack form factor allows
two xSwitches to be racked side-by-side, or placed in a rackmount with Telos Alliance xNode IP-Audio
interfaces. Noiseless and fan-free; can be conveniently placed adjacent to your audio devices, rackmounted using included hardware, or wall-mounted (with an accessory kit available separately).
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FEATURES
• Fanless design with heavy cast-aluminum heat-sinks is completely silent in-studio. Front-panel heat
sinks are cooled by ambient air, not “rack air,” eliminating overheating worries.
• Friendly OLED front-panel display with port status, IP address, PoE status and operating temperature
readouts - features not available in other switches in this class.
• One-button setup eliminates programming and saves hours of setup time.
• Functions as a core switch for standalone studios, or as an edge switch in larger facilities, or at your
Ethernet-connected transmitter site.
• Allows Axia network admins to add network ports economically, a la carte, instead of 24 or 48 at a time.
• xSwitch supports IGMP (Internet Group Management Protocol) Version 2, used to manage Multicast
group traffic (an essential part of Livewire’s intelligent audio routing system).
• xSwitch can handle up to 2,000 Multicast groups, and 2,000 ARP table entries, meaning it can’t run
out of bandwidth. (Other 8-port switches support only 250 groups.)
• Superior support for low-latency media streams, using four-level hardware strict priority QoS — other
switches have only one strict priority queue.
• Works with Axia iProbe network management software, allowing easy administration from your office
PC or remotely via WAN connection.
• Part of Telos Alliance’s xNode family, xSwitch can be used as a freestanding device or racked singly or
side-by-side with other xSwitch or xNode devices.
• Premium components include rugged cast faceplate and heat sinks, high-resolution OLED displays,
and bulletproof power supplies designed for high-availability telecom applications.
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TELOS ALLIANCE | XSWITCH
IN DEPTH
Your radio station needs programming.
Your network switch shouldn’t.
We invented Livewire in 2003, with the idea of saving money in broadcast studio construction by using
off-the-shelf Ethernet switches to power networks that distribute broadcast-quality audio nearly
anywhere — across the hall, across the building, or across town. Some said it would never work! But
10 years later, Axia is the #1 brand of IP consoles, networks and routing equipment to broadcasters
worldwide. Maybe it’s because Livewire IP-Audio is so flexible and easy to use that clients regularly
tell us of days – even weeks – shaved off of studio installation time with components that simply click
together using Cat-5 cables. Not to mention the money they’ve saved with Axia, compared to oldfashioned hard-wired studio builds.
But Axia fans told us there was one thing that could make Livewire even easier to install: A network
switch that doesn’t require setup or programming. So our engineers went to work. The result: xSwitch,
the world’s only zero-configuration network switch designed specifically for the needs of IP-Audio
broadcasting.
xSwitch is different from any other Ethernet switch, because it’s custom-tailored to the needs of
Axia Livewire users. You see, third-party switches – even those certified for use with Axia – require
programming to correctly configure them with the QoS settings Axia networks demand. Which generally
means connecting a PC to the switch with a special cable, downloading a terminal emulation program,
and entering lines of parameters and instructions.
Perhaps you’ve already got an Axia network installed (thank you!). Will an xSwitch work with the Axia
gear you already have? Naturally! xNodes speak Livewire, the AoIP protocol that powers more than
50,000 networked pro audio devices at radio and TV stations around the world. One click to hook up, and
they’re ready to go.
xSwitch does away with switch programming. Our experts have already pre-configured xSwitch with all
the instructions needed to run Livewire perfectly, flawlessly, out of the box. All you have to do is plug it
in, perform a quick one-button setup, and start connecting Livewire devices. Easy, yes?
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TELOS ALLIANCE | XSWITCH
On the xSwitch’s connection panel, you’ll find two SFP (Small Form-Factor Pluggable) Gigabit ports, in
addition to dual 1000BASE-T copper ports. Use the SFP ports for copper or fiber connections to your
Livewire network. The adjacent 1000BT copper ports provide a dual-redundant network interface; if
the primary network link is interrupted, the secondary backup connection is automatically activated.
You’ll also find 8 100-BaseT Livewire ports, 4 with PoE (Power over Ethernet) to power xNodes audio
adapters, Telos VSet telephones, or any other network device that uses the IEEE 802.3af standard.
Speaking of power, note the internal, auto-ranging power supply with professional IEC connector: you’ll
never find wall-warts powering Axia gear.
xSwitch is built using the chassis developed for our award-winning xNode family of AoIP audio adapters,
the latest generation of half-rack, high-performance IP-Audio interfaces. They’re fanless, which means
they’re noiseless too; you can put them in any studio. They have a versatile mounting arrangement
that lets you deploy two xSwitches into just 1RU of rack space (or rack an xSwitch alongside an xNode).
This allows you the flexibility to do things impossible before — like combine an xSwitch with xNodes
to create a “Supernode”. An xSwitch connecting 8 analog xNodes creates a 32x32 stereo router - or a
64x64 mono router - in the space of just 4RU. Great for making an audio snake, for adding I/O to that
add-on studio on the next floor, or even as the heart of a standalone studio.
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TELOS ALLIANCE | XSWITCH
SPECIFICATIONS
Power Supply AC Input
• Auto-ranging supply, 95VAC to 240VAC, 1.0 A, 50 Hz to 60 Hz
• IEC receptacle, internal fuse
• Power consumption: 75 Watts (all PoE ports under load)
Power over Ethernet
• 15.4 W-per-port maximum, 61.6-W switch maximum
Environmental Ranges
• Operating temperature: 32° F to 104° F (0°C to 40°C),
• Relative humidity: <90% (noncondensing)
Physical Dimensions
• 8.5” (22 cm) wide; two may be mounted side-by-side in a standard 1RU rack space (with included
mounting kit)
• 1.72” (4.4 cm) height, 11.75” (30 cm) depth
• Shipping Weight: 7 lbs. (3.2 kg.)
• Shipping Dimensions: 17” (43.2 cm) length, 13” (33 cm) width, 7” (17.8 cm) height
Ethernet Switch Specifications
• 4 QoS levels
• VLANs supported: 1
• Hardware filter capacity: 8,000 (this is the total limit of MAC addresses + multicast group
count supported).
• Supported protocols:
• IPv4 hardware switching
• IGMP version 2 snooping
• IGMP snooping querier
• DSCP (IP Type Of Service based priority)
• 802.1p (Ethernet 802.1Q tag priority)
• HTTP (WEB based management)
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TELOS ALLIANCE | XSWITCH
• Ports 100BT 1, 2, 3, 4: 100BASE-T Fast Ethernet (10/100MBit/s), Power-over-Ethernet source
• Ports 100BT 5, 6, 7, 8: 100BASE-T Fast Ethernet (10/100MBit/s)
• Ports GIG 1, 2: 100BASE-T Copper or SFP (Small Factor Pluggable Transceiver) module
IGMP Snooping Parameters
• Router present time out: 400s
• Query Response Interval: 10s
Connector Specifications
10/100/1000 Ports
The 10/100/1000 Ethernet ports use standard RJ-45 connectors.
Connecting to 100BASE-T-Compatible Devices
When connecting the ports to 100BASE-TX-compatible devices, you can use a two or four twisted-pair,
Category 5e, straight-through cable.
Connecting to 1000BASE-T Devices
When connecting the ports to 1000BASE-T devices, you must use a four twisted-pair, Category 6,
straight-through cable.
SFP Module Ports
The SFP module slot on a dual-purpose port uses SFP modules for fiber-optic and copper uplink ports.
xSwitch works with the following supported SFP modules:
• Cisco Copper SFP Model:GLC-T=
• Cisco Copper SFP Model: SFP-GE-T=
• Cisco Multimode fiber model: GLC-SX-MMD=
• Cisco Multimode fiber model: GLC-SX-MM-RGD
Regulatory
North America: FCC and CE tested and compliant, power supply is UL approved.
Europe: Complies with the European Union Directive 2002/95/EC on the restriction of the use of certain
hazardous substances in electrical and electronic equipment (RoHS), as amended by Commission
Decisions 2005/618/EC, 2005/717/ EC, 2005/747/EC (RoHS Directive), and WEEE.
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AXIA | XSELECTOR
xSelector Router Panel
The Network Switch Built for IP-Audio
OVERVIEW
The Axia® xSelector combines the routing functions of an XY router control panel with the audio
outputs of an Telos Alliance® xNode. In addition to analog, AES3 and headphone outputs, the Router
Selector Node also features an analog and an AES3 input — ideal for production or news studios
where operators both create and play audio streams. Six convenient “radio buttons” can be quickly
programmed for instant access to favorite sources.
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AXIA | XSELECTOR
FEATURES
• Fanless design for silent in-studio operation.
• High-resolution front-panel multi-function OLED display meters inputs and outputs and provides
audio source selection controls.
• Local I/O connections via industry-standard RJ-45 or XLR audio connectors,.
• In addition to being able to select audio streams from the Livewire® network for use locally, the
xSelector features one stereo input and one stereo output, allowing fast network distribution of locally
created streams from audio workstations or portable audio devices. Each xSelector can create 1
stereo Livewire stream, which becomes available to other devices on the Livewire network.
• Local I/O is presented on both AES digital and analog balanced inputs and outputs. The user can feed
audio into either a balanced analog input or an AES input.
• Both the AES and analog outputs are active simultaneously; both outputs have the same audio
present.
• Includes two GPIO closures presented on standard DB-15 connectors for machine control of
associated devices.
• xSelector’s stereo outputs can be assigned to output either the locally-created audio stream, or a
single stereo Livewire stream acquired from the network and easily selected from a list of available
streams using the front panel OLED display.
• Six frequently-used streams can be assigned to the front panel “radio buttons” for instant access.
Filmcap buttons can be labeled with names of assigned channels if desired.
• Dual Livewire 100BASE-T Ethernet ports for redundant connection to your Livewire audio network.
• Front panel headphone jack and volume control make xSelector a valuable addition to dubbing and
ingest stations where minimal infrastructure is desired.
• Built-in HTTP server for easy remote control using any PC with a Web browser.
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AXIA | XSELECTOR
IN DEPTH
The production-room powerhouse.
The Axia xSelector looks a lot like a traditional XY router control panel, but it’s much, much more. So
much more, in fact, that you’ll make xSelector a staple in your TOC, production rooms, news stations —
anywhere your talent needs to both create and consume networked audio streams.
xSelector lets talent select from all available audio streams on the Livewire network, and route them to
its local output (conveniently presented in both balanced analog and AES/EBU format). xSelector is easy
to use: The front-panel LCD screen lists available network sources; talent uses the adjacent selector
knob to browse sources and then pushes the knob to “take” the selected source, instantly routing
that source to the local outputs for use with an audio workstation, a specific console input, recording
device, etc.
Also on the front panel, six film-cap “radio buttons” provide instant access to frequently-used
sources. There’s also a stereo ¼” TRS jack with a volume control which supplies an internally-amplified
audio output directly to talent headphones, making xSelector a perfect choice for small workstation
environments by eliminating the need for an external headphone amp.
Around back, you’ll find separate left and right balanced XLR and RJ-45 connections for the analog
input and output, another set of XLR and RJ-45 connectors for the AES/EBU input and output, DB15 connectors for the two GPIO machine-logic controls, and RJ-45s for the two redundant Livewire
100BASE-T Ethernet connections.
All this functionality makes xSelector the perfect choice for news booths or dubbing stations where only
one active feed is required, or for intake stations that allow non-technical folks to easily move audio
from external sources (like field recorders) into the Axia network.
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AXIA | XSELECTOR
SPECIFICATIONS
Connections:
AES/EBU
• 1x Stereo Input, presented on one XLR-F connection and one RJ-45 connection
• 1x Stereo Output, presented on one XLR-M connection and one RJ-45 connection
Analog
• 1x Stereo, presented on two XLR-F connections and one RJ-45 connection
• 1x Stereo, presented on two XLR-M connections and one RJ-45 connection
GPIO
• 2x DB-15, each with 5 opto-isolated inputs and 5 outputs
Network
• 2x 100BASE-T connections, presented on RJ-45
Audio:
Analog Line Inputs:
• Input Impedance: >40 k Ohms, balanced
• Nominal Input Range: Selectable, +4 dBu or -10dBv
• Input Headroom: 20 dB above nominal inputMeta
Analog Line Outputs:
• Output Source Impedance: <50 Ohms balanced
• Output Load Impedance: 600 Ohms, minimum
• Nominal Output Level: +4 dBu
• Maximum Output Level: +24 dBu
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Digital Audio Inputs And Outputs
• Reference Level: +4 dBu (-20 dB FSD)
• Impedance: 110 Ohm, balanced
• Signal Format: AES3 (AES/EBU)
• AES3 Input Compliance: 24-bit with sample rate conversion
• AES3 Output Compliance: 24-bit
• Digital Reference: Internal (network timebase) or external reference 48 kHz, +/- 2 ppm
• Internal Sampling Rate: 48 kHz
• Input Sample Rate: 32 kHz to 192 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
Frequency Response
• Any input to any output: +/- 0.5 dB, 20 Hz to 20 kHz
Dynamic Range
• Analog Inputs to Analog Outputs 108dB referenced to 0dBFs, 111dB A-weighted
• Analog Inputs to Digital Outputs 110dB referenced to 0dBFs, 113dB A-weighted
• Digital Inputs to Analog Outputs 112dB referenced to 0dBFs, 115dB A-weighted
• Digital Inputs to Digital Outputs 126dB
Total Harmonic Distortion + Noise
• Analog Input to Analog Output: < 0.005%, 1 kHz, +18dBu input, +18dBu output
• Analog Input to Digital Output: < 0.004%, 1 kHz, +18dBu input, -6dBFs output
• Digital Input to Analog Output: < 0.004%, 1 kHz, -6dBFs input, +18dBu output
• Digital Input to Digital Output: < 0.0003%, 1 kHz, -20dBFs
Crosstalk Isolation, Stereo Separation And Cmrr
• Analog Line channel to channel isolation: 90dB minimum, 20Hz to 20kHz
• Analog Line stereo separation: 85dB minimum, 20Hz to 20kHz
• Analog Line Input CMRR: 80dB minimum, 20Hz to 20kHz
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AXIA | XSELECTOR
Power Supply Ac Input
• Auto-ranging supply, 90VAC to 240VAC, 30 Hz to 60 Hz, IEC receptacle, internal fuse
• Power consumption: 35 Watts
Operating Temperatures
• 0 degree C to +40 degree C, <90% humidity, no condensation
Regulatory
North America: FCC and CE tested and compliant, power supply is UL approved.
Europe: Complies with the European Union Directive 2002/95/EC on the restriction of the use of certain
hazardous substances in electrical and electronic equipment (RoHS), as amended by Commission
Decisions 2005/618/EC, 2005/717/ EC, 2005/747/EC (RoHS Directive), and WEEE.
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AXIA | ROUTING CONTROL PANELS
Axia® Routing Control Panels
Fingertip control, just where you need it.
OVERVIEW
Axia makes it easy to build large, IP-based routing networks of up to 10,000 audio streams. We also
make hardware tools to help your operators control all that networking power. Along with PathfinderPC
and PathfinderPRO routing control software, Axia Routing Control Panels put fingertip control of inputs,
outputs and routing scene changes anywhere you’ve got a rack space.
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AXIA | ROUTING CONTROL PANELS
FEATURES
• Six accessory control panels for convenient talent / guest control of a variety of studio operations:
routing scene changes, GPIO closures, XY control of inputs and outputs, etc.
• Slim panel design mounts in any 1RU rack space.
• Fanless, convection-cooled.
IN DEPTH
Take Control.
Axia control panels let you place routing power anywhere — in a studio turret, a TOC control panel or
an equipment rack. These accessory control panels work with Axia’s PathfinderPC and PathfinderPRO
routing control hardware, allowing you to map routing commands – from simple contact closures to
complex logic-driven events – to any button for fast execution.
Film-cap controllers with LED-backlit keys can be illuminated with a choice of colors; keycaps are filmlegendable for quick function identification. SmartSwitch panels have dynamic, backlit LCD buttons that
can change color and text with user activation. And the rack-mount 8-button SoftSwitch panel has highresolution OLED buttons that can be loaded with user-created bitmaps for instant function identification.
And the XY Routing Control Panel allows convenient on-the-fly routing of networked sources from
anywhere in your facility; route any source to any output of your choosing with just a couple of knob twists.
17-Button LCD SmartSwitch Panel
The 17-button SmartSwitch Router Control Panel features backlit LCD buttons with dynamic text and
color to provide 1-touch remote access to often-used machine-control or software functions. Multiple
pages of button assignments can be programmed and recalled with just a touch; use PathfinderPC’s
Stacking Events Editor to map single commands or complex routing salvos to any button. Easy-to-use
Web-based configuration pages can be accessed from any PC on the Livewire® network.
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AXIA | ROUTING CONTROL PANELS
Film-Cap Switch Panels
Use these Film-Cap Router Control Panels when dynamic-text capabilities are not required; lighted
aircraft-grade switches provide fast execution of router salvos, machine-control or software functions
programmed using PathfinderPC Router Control software. 5-, 10- and 15-button rackmount models are
perfect for use in a studio turret, TOC control panel or equipment rack. Place film labels under the clear
button caps; set the LED backlights to any of 8 different colors.
OLED SoftSwitch Router Control Panel
The 8-Button OLED SoftSwitch provides high-visibility router control from any studio turret or
equipment rack. Its eight bright, sharp OLED (Organic Light-Emitting Diode) readouts can display simple
text or user-supplied monochrome bitmaps, and can be seen from nearly any angle — across the table,
or across the room. Use PathfinderPC to program custom routing commands you can invoke instantly
with the touch of a button.
XY Router Control Panel
XY Router Control Panel lets you route any source to any destination (any-to-any routing) with the click
of a button. Choose your desired audio stream, select your network output and press “Take” to route
audio. Perfect for TOC program stream selection, ingest stations where a multitude of incoming feeds
need routing to air, or production rooms — anywhere you need many-to-one control of networked
audio streams.
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AXIA | ROUTING CONTROL PANELS
SPECIFICATIONS
General
• Rackmount package requires 1RU of free rack space.
• All Router Control Panels require 1 free 100BASE-T Ethernet port on a network switch for connection
to the Axia network.
• 17-button SmartSwitch Panel requires PathfinderPC or PathfinderPRO software to program and
execute conditional routing commands.
• FilmCap Button Panels require 1 free Axia GPIO port per each 5 buttons. PathfinderPC or
PathfinderPRO software is not required for GPIO command of networked devices, but is required to
program and execute routing commands.
• 8-Button OLED SoftSwitch requires PathfinderPC or PathfinderPRO software to program and execute
conditional routing commands.
Regulatory
North America: FCC and CE tested and compliant, power supply is UL approved.
Europe: Complies with the European Union Directive 2002/95/EC on the restriction of the use of certain
hazardous substances in electrical and electronic equipment (RoHS), as amended by Commission
Decisions 2005/618/EC, 2005/717/ EC, 2005/747/EC (RoHS Directive), and WEEE.
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AXIA | STUDIO CONTROL PANELS
Axia® Studio Control Panels
Give Your Talent The Power
Fusion Panels
Element Panels
OVERVIEW
Axia Studio Control Panels are a family of options panels designed for flush-mounting in desktop or
turret cabinetry. They allow you to place control of headphone source selection, mic off/on control and
even GPIO machine control at talent and guest desk positions, where they’re most convenient.
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AXIA | STUDIO CONTROL PANELS
FEATURES
• Six accessory control panels for convenient talent / guest control of frequently-changed options,
including headphone and mic control, GPIO closures, routing scene changes tied to Axia PathfinderPC
software, Talkback to CR board op or Guest postions.
• Easy RJ-45 connection to console CANBus control network.
• May be flush-mounted in any flat or vertical solid surface.
• All panels measure 6” x 2”; require 2” mounting depth.
IN DEPTH
Options are just a touch away.
Axia consoles are nearly synonymous with “flexibility.” You can save show settings and recall them in an
instant… customize backfeeds and routing salvos… share audio sources and control throughout your
facility… and that’s just the beginning. Axia helps you customize your studio too, with accessory control
panels that work seamlessly with your consoles to give talent fast access to headphone, mic and select
switching controls.
Mic Control Panel
The Mic Control panel gives talent or guests remote control of
their mic channel. Press the Talkback key, and you open a comm
channel to the board operator. There’s a handy Mute key for
those “frog-in-the-throat” moments, too. Works with all Axia
consoles.
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AXIA | STUDIO CONTROL PANELS
Producer’s Mic Control Panel
Designed especially to suit the needs of busy talk show
producers, the Producer’s Mic Control panel provides control of
microphone On/Off/Mute functions, and includes two special
Talkback keys so producers can easily converse with studio
remote talent. Works with all Axia consoles.
Headphone Selector Panel
The Headphone Selector panel lets talent control their own
headphone feeds. Turn the knob and control the volume. Push
the knob, scroll through the list of available sources, and push
again to “take.” Preset buttons are provided for instant access to
two programmed sources. Works with Fusion™ and Element®
consoles.
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AXIA | STUDIO CONTROL PANELS
Mic Control / Headphone Selector Panel
Why choose when you can have it all? Combination Mic Control/
Headphone Selector panel gives talent remote control of
headphone source and volume, mic channel on/off, and includes
Mute and Talkback functions. Works with Fusion and Element
consoles.
Five-Key Filmcap Button Panel
Five-key Button Panel can be placed wherever remote control of contact closures
or routing commands is desired. Film-legendable keys contain LED backlights with
individual color settings, and work with Pathfinder routing control software to put
fingertip control right where it’s needed. Works with Fusion and Element consoles.
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AXIA | STUDIO CONTROL PANELS
Four-Key SmartSwitch Button Panel
Four-key SmartSwitch has illuminated, dynamic LCD keys that
can change text and backlight color based on conditional logic
macros you construct in Pathfinder routing control software,
using simple drop-down tools. Works with Fusion and Element
consoles.
SPECIFICATIONS
General
• Desktop panels require access to Axia CANBus control network via CAT-5 connection.
• Flush-desktop mounting style requires routed 6” x 2” cutout in countertop or work surface. 2” of space
required behind each panel for adequate connector/cable clearance.
• Not all panels work with all Axia consoles. Consult Axia or your Axia representative for specific
applications.
Regulatory
North America: FCC and CE tested and compliant, power supply is UL approved.
Europe: Complies with the European Union Directive 2002/95/EC on the restriction of the use of certain
hazardous substances in electrical and electronic equipment (RoHS), as amended by Commission
Decisions 2005/618/EC, 2005/717/ EC, 2005/747/EC (RoHS Directive), and WEEE.
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AXIA | LIVEWIRE+ AES67 IP-AUDIO DRIVER
AXIA® LIVEWIRE+™ AES67
IP-AUDIO DRIVER
Pure Digital Audio from Networked PCs
OVERVIEW
The Axia Livewire+ AES67 IP-Audio Driver is the very first AES67-Compliant* IP Driver. It lets you send
and record single or multiple channels of stereo PC audio directly to and from Axia networks via Ethernet
— no sound cards needed. Up to 24 channels of stereo audio can be sent simultaneously over a single
CAT-5 Ethernet connection.
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AXIA | LIVEWIRE+ AES67 IP-AUDIO DRIVER
FEATURES
• First AES67-compliant IP-audio driver.
• Sends audio sources to the Livewire® / Livewire+ AES67 network from PC/Windows audio
applications such as multichannel delivery systems and other audio players.
• Receives audio from the Livewire / Livewire+ AES67 network to destinations on the PC/Windows
system, such as audio recording applications.
• GPIO function conveys “button-press” data from the Livewire /Livewire+ AES67 network to
destination applications; i.e., a console fader start button can command a PC/Windows-based audio
player to start playback.
• The Axia Livewire+ AES67 IP-Audio Driver single-stream version emulates a standard sound card, with
one stereo audio output device and one stereo audio input device. This version is suitable for typical
two-channel (stereo) playback or recording applications.
• Axia Livewire+ AES67 IP-Audio Multichannel OEM versions emulate 4, 8 or 24 stereo channel sound
cards (depending upon installed version), with one stereo audio output device and one stereo audio
input device per “sound card.” These versions are intended for more complex professional applications.
• Supports 5.1 surround audio streams as well as stereo, configurable on a per-stream basis.
• Windows version includes WDM and ASIO versions for maximum system flexibility.
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AXIA | LIVEWIRE+ AES67 IP-AUDIO DRIVER
IN DEPTH
Pristine PC Audio: No Sound Card Required
Way back when enormous cart machines still roamed the earth freely, we used XLR connectors to get
recorded audio into the console. But when PCs replaced the cart machine, we continued to connect to
their sound cards with plain-Jane XLRs and a thick bundle of discrete wires that can’t carry logic, PAD
or any of the useful information that PC playout systems provide. Why? With the Axia Livewire+ AES67
IP-Audio Driver, there’s a better way, and it’s now AES67-compliant. We’ve added the ability to sync to
PTP, support to define multicast address outside of the Livewire range, and added SIP/Unicast support
for RTP streaming.
The PC is the heart of the modern radio studio. And Axia makes it easy to connect and exchange pristine
digital audio with it. The Axia IP-Audio Driver for Windows is a special Windows driver that feeds your
digital audio directly from your PC’s Ethernet port to the Livewire / Livewire+ AES67 network. Up to 24
stereo playback channels and 24 stereo record channels can be accessed using our multi-stream driver
that’s provided by your favorite digital delivery system provider; a single-play/single-record version is
available for audio workstations.
The IP-Audio Driver also provides GPIO-like start/stop and other control functions over the same
network. It’s available with the latest versions of high-end Windows audio delivery and editing
software applications such as those from BSI, Burli, DAVID Systems, Dalet, ENCO, iMediaTouch, Netia,
RCS, WideOrbit, and Zenon Media (to name just a few) and for Linux-based Rivendell through Paravel
Systems—more than 50 systems and counting.
The Windows version of the IP-Audio Driver is available to broadcasters directly from Axia in 1-Stream
and 4-Stream versions, and from Axia Delivery System Partners in 8-Stream and 24-Stream versions.
Linux versions are available from our partner, Paravel Systems. For a full listing of Axia Delivery System
Partners, visit www.TelosAlliance.com/Partners.
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AXIA | LIVEWIRE+ AES67 IP-AUDIO DRIVER
SPECIFICATIONS
Microsoft Windows™ Operating System Requirements
• Windows 7 and Windows 7 Pro (32- and 64-bit editions)
• Windows 8
• Windows 10 (32- and 64-bit editions)
• Minimum hardware requirements specified for your Windows operating system are sufficient to run
the Axia IP-Audio Driver.
Linux Operating System Requirements
• The Axia IP-Audio Driver for Linux is sold exclusively through Paravel Systems. Please contact them at
ParavelSystems.com/contact-us/
* One of our goals at the Telos Alliance is to further the adoption of AES67, the standard for Audio over
IP designed to allow interoperability between various IP-based audio networking systems. And while
these two words may sound the same, the differences between AES67 compliance and compatibility
is huge. AES67, like all standards, can be minimally implemented. And when standards are minimally
implemented, they minimally get the job done. Simply put, compliance means that every single aspect
of the AES67 standard—like Unicast mode, for example—is met. Livewire+ AES67 is fully AES67compliant.
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AXIA | PATHFINDER
Pathfinder Routing Software
Routing Automation for Axia® Networks
OVERVIEW
Axia’s PathfinderPC and PathfinderPRO router control software for Windows is an amazingly rich set of
tools you can use to customize and command your entire Axia network, allowing you to craft extremely
sophisticated routing functions. Define automated switching events, construct custom software control
panels, change between presets manually, on a daypart schedule, or via an external trigger. Pathfinder’s
advanced features include the ability to sense silence at a particular audio port and patch around it
automatically — and even send the engineer an e-mail notification. And that’s just the start.
Designed for automated routing control in small to medium-sized facilities, PathfinderPC provides a
central point of control, via IP, of up to 25 Axia devices in your plant. Capabilities include route or scene
changes based on scheduled events, GPIO closure or Silence Detect trigger events.
PathfinderPRO is the enterprise version of PathfinderPC. It contains all features found in PathfinderPC
plus additional capabilities tailored to facilities with large physical plants or complex operational
requirements. PathfinderPRO Controls an unlimited number of Axia devices and supports unlimited
PathfinderPC client or PathfinderPC Mini connections plus direct Pathfinder control of motorized console
faders, VMix Virtual Mixers and more. Includes two server licenses for backup server or server clustering.
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AXIA | PATHFINDER
FEATURES
• Provides central routing interface for entire Axia networks. Pathfinder presents all Axia nodes and
equipment and as a traditional single router, so you don't need to jump from place to place to see and
manipulate your facility's routing infrastructure.
• Provides scheduling for one-time or regularly occurring routing events. These events can trigger route
changes and/or GPIO events, from small-scale changes to system-wide “scene changes”.
• Provides silence detection events with email and GPIO warnings, and automatically switches to an
alternate pre-programmed route in the event of an audio failure. Includes audio presence meters in
the onscreen routing matrix for instant visual audio confirmation.
• Pathfinder Stack Events allow you to design logic as complicated or as simple as you need, using simple
or complex Boolean based logic events. So if you need to route a specific satellite feed to air only on
Mondays, when a certain audio route exists, and the operator is holding down the blue button, you can.
• Panel Designer allows you to drop controls into your own software routing panel, then deploy it on studio
PCs for users to select routing changes, monitor silence detection, and a whole host of other functions.
• Interfaces routing control with Axia console User Keys. Map custom-designed features to buttons
mounted right in the console; depending upon console equipment, these buttons can even be
programmed to dynamically change color and text to display status and engage actions.
• Built-in Protocol Translator, a part of PathfinderPRO software, allows your legacy systems to think
they are talking to a router they understand — while behind the scenes, it's really Axia. Currently
supported third party protocols include Pro-Bel General Router and General Switcher Protocols, Sine
Systems ACU-1 Protocol and SA Port Router protocols.
• Clustering support with Pathfinder PRO allows deployment of multiple PathfinderPRO servers which
automatically monitor each other for backup and redundancy. If the primary server is unavailable, the
clustered backup takes over, and all the client screens in the studios follow.
• Comprehensive logging capabilities include every route change, GPIO change, user button press, and
more. Logging for each of these features may be enabled or disabled individually.
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AXIA | PATHFINDER
IN DEPTH
Power and Flexibility, At Your Fingertips
Power is good — but only if you can control it. Axia’s Pathfinder family of router control tools let you
customize and command your entire Axia network. Using your choice of graphical software or networked
appliance, you can easily build extremely sophisticated routing functions like automated events,
custom on-screen control panels — even change the entire network on a timed schedule if you like.
Pathfinder can even give you peace of mind, by sensing silence on critical paths and patching around it
automatically — then sending you an e-mail to let you know what happened. And that’s just the start.
Pathfinder can keep automatic logs of your studio network’s routing operations — route changes, GPIO
changes, user button presses, and more. Create sophisticated routing “scenes” with Boolean logic that
automatically watch for and react to specified events, using a unique graphical editor that eliminates
tedious script writing. Pathfinder Panel Designer even lets you construct custom on-screen controls that
can be deployed on PCs across your network. Or, map custom features to rack-mounted button panels
and user keys mounted right in the console.
In today's broadcast environment, information is key. So Pathfinder allows you to keep logs of your
studio network's routing operations — route changes, GPIO changes, user button presses, and much,
much more.
You want to design that perfect automated system? Pathfinder gives you the tools. Pathfinder Stack
Events allow you to design logic as simple (or sophisticated) as you need. An enhanced, graphical editor
eliminates tedious script writing, allowing you to create sophisticated routing "scenes" with Boolean
logic that automatically watch for and react to specified events.
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AXIA | PATHFINDER
Pathfinder's Panel Designer applet lets you construct custom on-screen controls that can be deployed
on PCs across your network. Or, map your custom designed features to rack-mounted button panels and
user keys mounted right in the console. Some of these buttons can even dynamically change color and
text to display status and engage actions.
There are two different Pathfinder software offerings tailored to your specific needs. Read on to find out
which is right for you.
PathfinderPC Software
Designed for automated routing in small to medium-sized facilities, PathfinderPC gives you networked
control of up to 25 Axia devices. This full-featured system runs on Windows PCs and allows you to
construct and execute route or scene changes based on scheduled events, GPIO closures or Silence
Detect trigger events. Using the client application, you can log in and change routing from anywhere
you have network or Internet access. Use PathfinderPC to attach events to Axia SmartSwitch,
SoftSwitch and Film-Cap button panels, or construct on-screen “virtual” controls that can run
simultaneously on up to 10 PCs.
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AXIA | PATHFINDER
PathfinderPRO Software
PathfinderPRO, the enterprise version of Pathfinder, contains all of the features found in PathfinderPC
plus additional capabilities tailored to facilities with large physical plants or complex operational
requirements. PathfinderPRO supports server “clustering” – running simultaneously on two connected,
yet independent computers – for the ultimate in redundancy and security.
PathfinderPRO catalogs all of your Axia devices and supports as many end-user connections as your
CPU can handle. PathfinderPRO can directly control console VMix virtual mixers, motorized faders on
consoles so equipped, Show Profile changes, and more.
But PathfinderPRO doesn’t stop at just controlling your Axia equipment. Complete delivery system
integration is at your fingertips with Sine Systems ACU-1, Pro-Bel, SA Port Router and generic protocol
emulators, plus support for routing and translating of serial, TCP and UDP ports. Snap-in real-time
metering and Web browser controls provide added options for user-designed software panels. Browser
controls even support multimedia audio and video, allowing embedded A/V streaming displays in
software mini-panels.
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AXIA | PATHFINDER
SPECIFICATIONS
PathfinderPC Server
Hardware
• Minimum hardware requirements specified for Windows XP, 2003 Server, 2008 Server, and Windows
7 are also acceptable to run PathfinderPC Client, PathfinderPC Mini, Panel Designer, SAPortRouter,
VMIXControl, and the bridge application programs.
Software
• Windows XP, 2003 Server, 2008 Server, or Windows 7/8. Microsoft .NET 3.5 SP1 is also required.
Additionally, Windows 7 and 2008 require that the startup links be set to “Run as Administrator” in the
compatibility frame.
PathfinderPRO Server
Hardware
• Minimum 512 Mb RAM, If the clustering option is used, minimum two NIC cards should be used, four
are recommended.
Software
• Windows XP, 2003 Server, 2008 Server, and Windows 7/8/10. However, installations using more
than 10 clients (PathfinderPC Client, PathfinderPC Mini) will require a server operating system such as
Windows 2003 server or Windows 2008 Server.
PC Client Applications
• Windows XP or better, or any workstation
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AXIA | PATHFINDER CORE PRO
Axia® Pathfinder Core PRO
Routing Automation &
Facility Management Appliance
OVERVIEW
Experience the Power of Simplicity
Introducing the second-generation of Axia Pathfinder, the Pathfinder Core PRO Advanced Routing
Control Appliance. Never before has routing control been this simple...or this sophisticated. Axia has
taken everything we’ve learned over the last 10 years since we launched the Pathfinder family of routing
control software and ported that experience into a completely new product, designed from the ground
up to provide pro-grade routing control that is shockingly simple to use, taking routing control and facility
management to a new level.
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AXIA | PATHFINDER CORE PRO
Pathfinder Core PRO is a toolbox with numerous features and uses that combine to create more efficient
workflows and facility management. For one, it centralizes your Axia devices, letting you make route
changes from one central place—the Pathfinder Core PRO web interface. Pathfinder Core PRO simplifies
routing in complex facilities with hundreds of sources and hundreds of audio destinations by giving the
user easier, more intuitive control over audio workflows.
This second-generation of Pathfinder routing control lets you customize and command your entire Axia
network like never before with streamlined functionality including Logic Flows—a new flow-chart-style
events system that makes events easier to create, adjust, and monitor in real time. An efficient, intuitive
web interface means easy configuration and monitoring from any device. Finally, this Linux-based
appliance makes better use of the processor, while freeing you from a Windows-based server.
A Word About Axia
Axia is the AoIP division of the Telos Alliance, technology leader in professional broadcast audio
equipment for radio broadcasters since 1984. In 2003, we introduced the world’s first Ethernet-based
console system for broadcasting with Livewire, an AoIP protocol that enables high-reliability, low-delay
uncompressed digital audio over Ethernet, including audio, logic, control, and program associated data
(PAD). At the time this was a new concept, but VoIP showed the telecom industry how powerful, flexible,
and cost-efficient it was to move audio via IP, and the idea caught on fast.
Since those early days, Axia and the Telos Alliance have been committed proponents of AoIP
interoperability—first as charter, supporting members of the X192 Working Group that defined AES67,
and now as founding members of the Media Networking Alliance, a group of prominent equipment
manufacturers formed to actively promote the adoption and implementation of the AES67 standard.
Today, 7000+ Axia mixing consoles and 70,000 Livewire connected devices—along with those of 85+
Livewire partners—are powering broadcast studios around the globe. (For a complete list of Livewire
partners, click here. ) And the list keeps growing.
Axia helps broadcasters build studio facilities to meet today’s most demanding applications. With Axia,
you can quickly and easily connect a few rooms, or an entire facility. Axia networks have a total system
capacity of more than 10,000 audio streams, and can carry hundreds of digital stereo channels (plus
machine logic and PAD) over a single CAT-6 cable, eliminating much of the cost normally associated with
wiring labor and infrastructure.
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AXIA | PATHFINDER CORE PRO
For example, a couple of Telos Alliance xNodes, connected together, can move a group of audio signals
over an Ethernet cable from one room to another. Connect with fiber and you can go across campus.
Attach a few more nodes and a switch and you have a distributed multi-room routing switcher. Plug
in a mixing surface and console engine to add a powerful networked broadcast console. Add intercom
stations for broadcast-quality plant communications that can be taken to air. Plug in your delivery
system PC and you can transfer files, live audio, and associated data all over the same network. And
since Axia audio is networked, analog and digital signals are merged seamlessly; cross-point switching
from any source to any destination is fast and simple. Then use Pathfinder Core PRO to bring all of our
routing together in one easy-to-use interface. Logic Flows within Pathfinder Core PRO react to button
presses, silence alarms, time of day commands and other system-wide events to generate alerts and
automate the workflow of the entire system.
But there’s much more to Axia than just the network. Once all of your consoles, peripheral devices and
computer workstations are connected together for unlimited sharing, it’s a cinch to add phone systems,
audio processors, codecs, satellite receivers, program delay units, or any audio device from the evergrowing list of Axia Partners. All of these devices work together in tight integration, which leads to more
intuitive and intelligent operation. By taking advantage of network efficiencies, Axia simplifies, saves you
money, gives you choices, and prepares your studio for the demands of today — and tomorrow.
FEATURES
• Reliable, redundant, system-wide routing control
• PC-platform independent; you can use almost any device to interface
• Linux-based network-attached appliance with a web user interface provides route control and
customized logic events
• Graphical interface with real-time state reporting
• New logic gates for creation of complex logic
• Control protocol for third-party integration, including Device Emulators
• Improved Memory Slot functionality, making this feature more powerful than ever
• Automatic router table generation
• Fan-free for cool, silent in-studio deployment
• Equipped with dual Gigabit Ethernet ports and dual-redundant internal power supplies
• Virtual routing
• Customizable user panels
• Includes 500 crosspoint and 500 Logic Flow endpoints
• Add-on licenses for additional 500 crosspoints or 500 Logic Flow endpoints available
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AXIA | PATHFINDER CORE PRO
IN DEPTH
1.
Telecom-grade dual, redundant, auto-switching power supplies are engineered for bulletproof reliability.
2.
Power supplies are unitized for quick replacement in the field. Just slide out the old, slide in the
new.
3.
Separate WAN jack provides a secure, separate connection to the outside world - make
routing changes from home, office or on the road.
4.
Pathfinder Core PRO controls your complete Axia routing network with one simple CAT-5
connection.
5.
Your rack has enough noisy stuff in it. Pathfinder Core Pro isn’t. Fanless convection cooling
means whisper-quite operation.
PC-Free
The Axia Pathfinder Core PRO appliance is ideal for stations that want reliable, redundant, systemwide routing control that is independent of a PC/Windows Server. The Linux-based network-attached
appliance for routing control includes a web user interface to provide route control and customize logic
events. Because it uses a web interface, Pathfinder Core PRO becomes PC-platform-independent;
users can use almost any personal computing device to interface with the appliance for reliable, 24/7
unattended operation. In addition, the traditional client applications our users have come to know and
love also work with Pathfinder Core PRO.
Logic Flows
Pathfinder Core PRO features Logic Flows, an all-new events system that uses logic gates for the
creation of complex logic and dramatically simplifies route changes. The Logic Flows event system
features a graphical flow-chart style that allows you to see the connections and manipulate them easier,
as well as see state changes in real time via the web interface. For example, if an endpoint changes, you
will see those changes live via a color and/or textual change on the Logic Flow graphic. Logic Flows are
intelligent enough to sense the connections between different flows and present those connections
graphically with a much larger array of properties that are available for use. Finally, we have added the
ability to clone entire groups of object settings with one simple flow, which becomes invaluable when
creating redundant logical paths. These are dramatic improvements over previous versions.
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Examples of PDM Logic Flows.
Example of airchain switcher requiring dual button press for safety.
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Device Emulators Control Protocol for Third-Party Integration
Because there are a variety of automation systems that don’t understand the Axia language, Pathfinder
Core PRO Device Emulators allow the unit to “look like” another protocol, emulating third-party protocols
and translating them into the Axia language. When Pathfinder Core PRO receives commands in a
non-Axia protocol language, it then makes the requested changes to the Axia equipment, emulating
a different kind of device by speaking that language. Additionally the Device Emulators allow the user
to define their own commands to send and receive, so Pathfinder Core PRO can be used to control and
receive commands from third-party equipment in that way as well. The user simply has to define the
textual commands to watch for or send in the Device Emulator.
Easy Setup
Pathfinder Core PRO is fast, efficient, and simple to use: Just attach to your network, give it an IP
address, and it automatically detects your Axia audio sources/destinations and GPIO ports. Pathfinder
Core PRO searches for and finds all of the Axia equipment on the network, then automatically creates
both an audio router and a GPIO router with all of the audio or GPIO sources and destinations that are
discovered within that equipment, simplifying the initial configuration.
Pathfinder Core PRO also gives you peace of mind by providing distributed redundancy within your
network. Multiple Pathfinder Core PRO units can be “clustered” for automatic redundant backup, and
each fan-free unit has field-replaceable dual-redundant power supplies as well. As a dedicated hardware
appliance, Pathfinder Core PRO gives you freedom from concerns about software compatibility,
automatic OS patches, and computer hardware limitations.
Customizable User Panels
In addition to advanced routing control, Pathfinder Core PRO lets users create their own user panels,
giving them a unique and customized look and design for features like meters and buttons.
Virtual Routing
Aside from routing physical inputs and outputs, Pathfinder Core PRO lets you create virtual routers. For
example, if a specific room in a facility doesn’t need to see the entire network of inputs and outputs, but
only the ones relevant to that room, then a Pathfinder administrator can create a virtual router that only
includes the sources and destinations required by that room. In addition, virtual routers can be used to
marry multiple sources and destinations together into routable packages. For example, you can create
a virtual source that has both an actual audio source and an actual GPIO source as part of its package.
Then when you create a route with the virtual router, the audio and GPIO get routed together with one
route change.
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Memory Slots
Memory Slots allow you to store—and make changes based on—custom pieces of information.
Memory Slots have been expanded in Pathfinder Core PRO to provide easy latching capability of any
property in the system, to follow the state of any property in the system, and to build custom text by
combining values from multiple slots together. This makes the use of Memory Slots more powerful
than ever.
500 Crosspoints, 500 Logic Rules
With Pathfinder Core PRO, you get 500 sources and 500 endpoint change blocks (within Logic Flows).
Need more? You can simply buy an extension license to add more logic or more crosspoints. Note: There
are a variety of crosspoints that don’t count against the license, including GPIO and virtual-only ones.
Only Axia audio sources count against the crosspoint license.)
FAQs
How is Pathfinder Core PRO different from the Pathfinder software offerings
from Axia?
Pathfinder Core PRO is a completely new product. It’s the second-generation of Livewire control ideal for
stations that want reliable, redundant, system-wide routing control that is independent of a PC. It has
several exclusive new features, like a new graphical web user interface with real-time state reporting
and logic gates for creation of complex logic, all in a 2RU appliance. Generally speaking, Pathfinder Core
PRO is easier to use, with more sophisticated and streamlined routing control over previous iterations.
Is Pathfinder required to run my Axia network?
No, Axia networks are self-contained routing and mixing systems that don’t require any external control.
However, if you want to automate your routing switcher, with preset scene changes, conditional routing
changes, or scheduled route changes, Pathfinder will satisfy your needs.
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Why would I need routing automation?
Pathfinder Core PRO lets you consolidate control of your network operations, bringing all of your
equipment together under one simple interface to create the most flexible workflows imaginable. It
takes all of your Axia nodes and equipment and presents them as if they were a traditional single router,
so you don’t need to jump from place to place to see and manipulate your facility’s routing infrastructure.
Can I trigger routing changes from studio consoles?
Yes. There are a variety of drop-in modules for our popular Axia Fusion consoles that you can use to
change single routes, or execute predefined salvos based on time-of-day or GPIO. Rack-mount panels
make it convenient to map Pathfinder routing commands to hardware buttons located elsewhere, too—
like your TOC, engineering office, or communications room.
Can Pathfinder sense dead air?
Yes, there’s a Silence Detect function, including Silence, Audio Presence, and Clipping on
Axia sources and destinations. With it, you can pick any audio stream in your network—
say, your Program-1 output—and monitor it for signal loss. Once your “silence” condition
is met, Pathfinder can take action by switching to a different audio input, flashing an alert
to talent, or sending you an e-mail.
Can Pathfinder react to a system command? I’m thinking about EAS activation…
Sure. You can define Logic Flows, in which Pathfinder Core PRO monitors GPI channels for external
commands and takes action when predefined conditions are met. So you could use a logic flow to
watch the GPIO output of your EAS decoder, and switch your main program output, along with your HD
channels, to be fed by the output of your EAS gear until the GPI is released—upon which audio inputs
are returned to their normal sources.
Additionally Pathfinder Core PRO provides generic Device Emulators that allow you to define custom
commands to send and watch for on a tcp port. So your automation system can send any command you
desire and Pathfinder Core PRO can detect that command, act upon it with any action you see fit via a
Logic Flow, and reply back with another custom command.
Finally, Pathfinder Core PRO provides an advanced control protocol that exposes everything you might
want an outside system to control.
Does Pathfinder Core PRO support clustering, like PathfinderPRO software?
Yes. You can attach two Pathfinder Core PRO devices to your network for complete routing automation
redundancy.
Does Pathfinder Core PRO support SNMP, AES70, or other protocols?
Pathfinder Core PRO supports Axia and protocols to allow you to create highly customized
workflows. As the industry is learning the importance of common tools, open protocols are becoming
hot topics. Pathfinder Core PRO was designed with an open architecture to easily incorporate other
protocols. Pathfinder Core PRO includes Device Emulators (protocol translators) with multi-protocol
support that can be used to send and receive any data.
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SPECIFICATIONS
Power Supply AC Input
90VAC to 240VAC, 50Hz to 60Hz, IEC receptacle, internal fuse
Power consumption: 100 Watts, auto-ranging
Operating Temperatures
-10 degrees C to +40 degrees C, <90% humidity, no condensation
Dimensions
W19.00in (48.0 cm), H 3.47 in (8.80 cm), D 10.40 in (26.4 cm)
2RU
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AXIA | IPROBE
Axia® iProbe
IP-Audio Network Management Software
OVERVIEW
iProbe software helps with management, updating, and remote control of any Axia system. It has a
powerful auto-documentation feature that generates configuration docs for every device, an Organizer
that allows grouping networked audio devices into logical groups for easy management, facilitates
uploading software to single or multiple devices, makes device configuration backups and more.
FEATURES
• Discovery: the ability to scan your Axia Livewire® network for Control Surfaces, Nodes (AES, Analog,
GPIO), and Mixing Engines, as well as any Livewire devices from Axia Hardware Partners.
• Displays current firmware versions running on the all connected devices and gives you the ability to
update firmware remotely, one device at a time or in logical groups of similar devices.
• Displays all the devices in your Livewire system and allows you to browse directly to a selected device’s
Web-based remote control interface. There is no need to type the device IP address into your browser.
• Complete configuration backup capabilities of individual devices, or all devices within your
Livewire system.
• Integrated Syslog server for automated event logging.
• Auto-Documentation feature exports complete system data to a format of your choice for secure backup.
• Built-in iPlay module allows you to listen instantly to any channel on your network, and verify the
levels of a given source.
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IN DEPTH
Easy Network-Wide Backup, Update and
Documentation
Axia iProbe is an intelligent network maintenance and diagnostics suite that consolidates managing,
updating, and remotely-controlling your Axia system into one easy-to-use software application.
Axia networked audio devices are managed using a standard Web browser to view, configure, and
administer each device. iProbe helps simplify this process by scanning and collecting all the information
and presenting it a graphical interface. Along with this convenient central point of control, iProbe
gives you powerful system tools like the Organizer, which performs advanced tasks such as gathering
Livewire-enabled devices into logical groups for easy management and single-point administration of
group settings. iProbe also helps with software version control, making it simple to upload software to
single or multiple devices, back up device configuration, and more.
There’s a powerful Auto-Documentation feature that queries and documents configuration settings
for every networked Axia device — essential for administering large networks. Auto-Doc gives you
the ability to export your Axia system data into an HTML format or text for printing a hard copy of your
system configuration, or constructing a web page for future reference. You can also export to a tabdelimited text format for importing into other documents or spreadsheets. Exporting in XML format to
other applications is also available.
And of course, there’s a System Backup / Restore function that generates full-system backups which
can be used to restore your Axia network from bare metal, should the need arise.
Even with all this power, iProbe is simple to use. An intuitive graphical interface lets you browse a list
of similar devices, or click one-at-a-time on individual devices to make inspections or adjustments. You
can even listen to individual sources with the integrated iPlay functions, and check on audio levels of any
audio source.
SPECIFICATIONS
System Requirements
• Windows 2000 Professional, Windows XP Professional, Windows Vista (32- and 64-bit editions),
Windows 7 and Windows 7 Pro (32- and 64-bit editions), or Windows 8 operating system.
• 100BASE-T or higher wired network adapter.
• Internet access to enable device firmware downloads.
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Axia® iPlay
Network Stream Player for Windows
OVERVIEW
Software-based IP-Audio monitoring program lets Windows PC users select and listen to any audio
source available to their Axia network. Choose from a complete list of available streams; eight userprogrammable preset buttons provide quick access to frequently-accessed channels. On-screen level
display meters auditioned audio.
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FEATURES
• Allows listening to any Axia network audio stream using standard PC sound card/speaker combo or
headphones.
• Automatic detection of Livewire® audio sources from connected PCs.
• User friendly interface allows filtering and sorting Livewire channels, making it easy to navigate in big
systems containing hundreds or thousands of Livewire channels.
• Preset buttons that allow quick access to eight pre-selected channels.
• Administrator can restrict access to a set list of audio channels using built-in Access Control Lists.
IN DEPTH
Turn Any PC Into a Listening Station.
Remember the days when giving your Sales Manager a listening station meant running cable through
the ceiling, installing a selector panel in the office wall, and mounting speakers in the drop tiles? And
then, he could only hear a limited number of the audio channels your plant produced.
Axia iPlay PC software does away with old-fashioned speaker wire and rotary selectors. It allows any
Windows PC to listen to streamed audio directly from your Axia network — any streamed audio. Not just
Program feeds, but satellite downlinks, remote hosts, news production studios, interview rooms, etc.
iPlay lets you give PC monitoring capabilities to PDs, GMs and sales staff using their existing computers,
with no special wiring required. Just connect their PC’s NIC to your Axia network, install iPlay, and presto!
Every PC is a listening station. There’s even an on-screen level display that meters the audio you’re
listening to — great for use as confidence meters for PDs or production personnel.
Users can choose from a list of all available audio, but in big plants that can be a lot to sift through. Not
to worry: they can pre-set their favorite channels on the eight user-programmable preset buttons to get
quick access to the streams they listen to most. And of course, you can filter out raw mic channels or
other selected audio streams to prevent unauthorized listening.
SPECIFICATIONS
System Requirements
• Windows 2000 Professional, Windows XP Professional, Windows Vista (32- and 64-bit editions),
Windows 7 and Windows 7 Pro (32- and 64-bit editions), or Windows 8 operating system.
• 100BASE-T or higher wired network adapter.
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iProFiler
Automated Program Archiving
OVERVIEW
Axia’s popular iProFiler logging software lets you simultaneously capture up to 24 stereo audio channels
to time-stamped MP3 audio logs directly from your Axia IP-Audio network — no audio cards required.
Included software records, manages and plays back archived audio files. Recording software runs under
Windows XP and later; playback software runs under Windows NT, Windows 2000, Windows 98 or
Windows XP and later. Record mode can be set for logging, skimming, or combination of both. Logged
audio may be auditioned remotely via LAN, WAN, or Internet.
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FEATURES
• Simultaneously captures up to 24 channels of stereo audio.
• Directly records Axia digital audio streams — no sound card needed.
• Archived audio can be auditioned remotely via LAN, WAN or the Internet.
• iProFiler Live Player streams audio over any IP connection as it’s being encoded. Great for consultants
or group PDs listening remotely.
• NTP Time Sync synchronizes log file timestamps with your house NTP server (if equipped).
• Choose your skimming mode: Logging (continuous archival storage of program material), Skimming
(records only when talent mic is open), or SmartSkimming (low-bitrate logging switches to a userspecified higher bitrate for quality captures when talent is on-mic).
• No “spool-up” time: iProFiler buffers incoming audio so that you never lose a word - no matter how
late talent opens the mic.
• Recorded audio is time-stamped and stored in easy-to-search 15 minute blocks for fast retrieval.
• Standard MP3 file format allows logged audio to be played back on any media player application. Play
files in iProFiler Archive Player to view detailed time-of-day data and user annotations.
• Easily select & export audio segments to WAV files for external editing.
• Choose any standard MP3 bit rate - from 16kbps - 320kbps - for the quality/drive space ratio that
best suits your needs.
• Encoded program segments can also be set to upload automatically to an external drive, network
share or FTP site.
• Remote monitoring application lets you “check up” on iProFiler remotely using a LAN or Internet
connection; monitors disk space & audio presence.
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IN DEPTH
Multi-channel, Multi-stream audio archiving for Axia Audio
networks.
Sooner or later, someone’s going to ask for a hard copy of a specific broadcast. Whether it’s a client
looking for proof of play, a Group PD that wants airchecks, or a listener claiming your morning show did
something naughty, you’ll need a record of your broadcast programming.
Be prepared with Axia iProFiler, the award-winning audio archiving software that integrates with Axia
IP-Audio networks to capture up to 24 simultaneous channels of Livewire® Standard-stream stereo
audio without sound cards. iProFiler uses the Axia IP-Audio Driver to exchange audio directly with Axia
networks; just install iProFiler on a PC, connect the computer’s NIC to the network with CAT-5, select the
program streams you want to capture. It’s as simple as that.
iProFiler’s networked connectivity makes it the easiest logger to set up and operate, bar none. Just
browse the audio streams available on your Axia network, select the ones you want to record, choose a
bit rate for storage and off it goes.
And iProFiler is extremely flexible; you can continuously log program audio, automatically record
telescoped talent airchecks, or record only what’s broadcast when the mic isn’t open. And iProFiler has
“listen line” capability that lets you hear audio over your network (or the Internet) as it’s being encoded perfect for group PDs or consultants.
iProFiler’s stored audio is networked, too. Any workstation or computer connected to your IP-Audio
network can find and listen to time-stamped audio using a simple web-browser interface.
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iProFiler gives you a choice of operating modes for each archived audio stream:
• Choose “Logging” for continuous archival storage of program material, indefinitely (dependent upon
storage space) or on a timed-record basis.
• Choose “Skimming” to record audio only when talent’s mic is open, to capture live shows, call-in
segments, talk shows or DJ bits. Program audio is pre-buffered so that there are no “up-cuts” upon
record activation.
• Choose “SmartSkim” for a unique combination of skimming and logging. When talent mics are closed,
ProFiler records audio in a low-bit rate logging mode, then switches to a higher bit rate for quality
captures when talent is on-mic. All bit rates are user-selectable.
iProFiler is ideal for stations required by law to log program content, and since you can also listen to
“live” audio over IP as it’s being logged, it’s great for Production Directors and morning show producers,
program consultants or group PDs. Perfect for competitive monitoring, too — log other stations along
with your own to fine-tune your formatics. An integrated audio browser lets your production crew tag
segments and export them as WAV files for further editing, and logged shows can be automatically
uploaded to FTP servers for storage or distribution.
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SPECIFICATIONS
PC Hardware Minimum Requirements
• Pentium-IV, 2.4GHz processor or better with 512Mb RAM, 300 Gb free hard drive space, 100BASE-T NIC.
Operating System
• iProFiler Server: requires Windows XP or later. WAN/Internet connection required for remote monitoring.
• iProFiler Client: Requires Windows XP, Windows Vista, or Windows 7 or later.
Operating modes:
• Logging (continuous archival storage of program material)
• Skimming (records only when talent mic is open)
• SmartSkimming (low-bitrate logging switches to a user-specified higher bitrate for quality captures
when talent is on-mic)
• Scheduled recording (date and time + length of program)
Audio Interface
• 100BASE-T or better Ethernet NIC with connection to Axia IP-Audio Network.
• Supports up to 24 stereo streams simultaneously.
Audio Specifications
• Storage Format: MP3.
• Comproession Algorithm: Genuine Fraunhofer IIS
• Bit Rates Available: 8 kbps to 320 kbps, in standard increments
• Pre-roll and Post-roll Skim delay: up to 10 seconds, user-definable
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AXIA | IP INTERCOM
IP Intercom
Go ahead: talk amongst yourselves.
OVERVIEW
Axia® IP Intercom is the only broadcast intercom system that takes advantage of the ease and efficiency
of proven IP-Audio technology. Using a standard Ethernet backbone, IP Intercom saves on cost, space,
and installation time, and eliminates special plug-in cards altogether.
The advantages of IP and Ethernet – low cost, easy installation and maintenance, efficient infrastructure
– are well known. Installing IP Intercom is as simple as clicking together Ethernet gear! And of course it’s
easily scalable: plug as many stations into your switch as you want and add on from there. There’s no
expensive, hard-wired, custom-cable multi-pair infrastructure to deal with.
If you don’t have an Axia studio network, IP Intercom can still help you save money, increase efficiency,
and decrease the hard-wired infrastructure hassle It’s a stand-alone system with I/O that will
accommodate multiple mixing consoles. But if you do have an Axia system, you’ll get seamless console
integration that gives your operators benefits other systems can’t, like the ability to take broadcast
quality intercom audio directly to air, and feed IFB audio directly to intercom callers.
The IP Intercom system includes a variety of desktop and rackmount stations, a software Intercom
application that turns any PC into an intercom station, and drop-in modules for popular Axia
mixing consoles.
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FEATURES
• 100% digital system, end to end.
• Seamless integration between broadcast audio and communications channels. Full 20 Hz – 20 kHz
audio response allows intercom channels to be taken to air with no degradation of sound quality
• Stand-alone rack-mount, desktop and integrated Axia console modules are available for a turnkey
intercom installation.
• Program station presets and GPIO functions using any standard Web browser.
• Ethernet-based system has no central matrix or card-cage; is naturally scalable. Easily expand the
number of intercom stations as your facility grows by simply plugging in new stations.
• Intercom keypad can also dial outside phone lines (using an optional telephone hybrid).
• Analog I/O presented on both XLR and StudioHub-compatible RJ-45 connectors.
• Front-panel locking connections accommodate popular mini-mics and headsets.
• Add PCs to the system with SoftCom Intercom Station for Windows.
IN DEPTH
Imagine a digital intercom system with no central matrix.
Actually, don’t bother — we’ve built one. Axia IP Intercom saves on cost, space, and installation time,
and eliminates special plug-in cards altogether. It’s real plug and play that works every time — even
when you need to add a station, or reconfigure the ones you’ve got.
Everybody knows the advantages of IP and Ethernet – low cost, easy installation and maintenance,
efficient infrastructure. Thanks to its efficient Ethernet backbone, installing IP Intercom is a simple
single-click connection. Of course it’s easily scalable: plug as many stations into your switch as you want
and add on from there. Then start talking! And if you move to a new location, you can just pick up the
gear and take it with you — there’s no expensive, hard-wired, custom-cable multi-pair infrastructure
mess to deal with.
Don’t have an Axia studio network? That’s OK. You’ll still save money and increase efficiency by choosing
IP-Intercom; it’s a stand-alone system with I/O that will accommodate multiple consoles. But if you do
have an Axia system, you’ll get seamless console integration that gives your operators benefits other
systems can’t. For instance, you can take broadcast-quality intercom audio directly to air. And you can
feed IFB audio directly to intercom callers.
IP Intercom gives you unlimited full-bandwidth access to any studio, news or sports venue, office,
hallway, broom closet or wherever. Talk and listen to individuals or groups hands-free, with no echo or
feedback — IP Intercom features exclusive AEC advanced echo cancellation from Fraunhofer Labs (the
inventors of MP3), so there’s never any open-mic feedback during conversations. Ever.
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IP Intercom system is completely digital. Other intercom systems try to make you think they’re digital
by piping their analog signals over CAT-5 cables, but the last thing you need during a breaking story
or transmitter failure is hum and buzz getting between you and the guy you need to talk to. With IP
Intercom, there isn’t any.
So you’ve gotta be a genius to use it, right? Actually, anyone with an index finger can operate this system
with ease. The web interface makes setup simple. Sharp, high-contrast OLED displays are easy to read
from anywhere in the room. And our clever callback feature makes sure you’ll never miss a call, no
matter what you’re doing. There are also functions that allow talent to mute calls from other stations, to
make sure there’s never an interruption on-air.
IP Intercom comes in several rack-mount and desktop styles, plus drop-in modules for Axia Fusion™ and
Element® consoles. And our unique SoftCom software lets you turn any connected PC into an intercom
station! Just mix and match to build a system customized to your needs.
Rackmount Stations
IC.20 Rackmount Station
The IC.20 intercom panel features 20 station presets for quick contact with frequently-called stations.
Perfect for Master Control or TOC, the IC.20 includes a keypad and associated display for fast access to
stations system-wide, plus group talk and auto-answer functions. Keypad can also dial outside phone
lines (using an optional telephone hybrid). 2RU rackmount package features high-visibility 10-character
OLED (organic LED) displays, built-in speaker, front- and rear-panel mic connections, 4-pin locking
headset jack, analog I/O presented on both XLR and StudioHub-compatible RJ-45 connectors, GPIO
connection for speaker mute/dim and external line-status tallies, and an Ethernet jack for single-cable
network connection.
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IC.10 10-Station Intercom Panel
The IC.10 is a 10-station version of the IC.20 we talked about earlier. It has 10 station presets with
high-resolution OLED displays, a built-in speaker, front- and rear-panel mic connections, 4-pin locking
headset jack, analog I/O on XLR and StudioHub-compatible RJ-45 connectors, GPIO connection for
speaker mute/dim and external line-status tallies, and an Ethernet connection.
IC.1 10-Station Intercom Panel
The IC.1 is a cost-effective way to add intercom capabilities to any studio. It features 10 LED-backlit
film-cap buttons that are easily labeled with station names; like other IP Intercom station, programming
is via Web interface. IC.1 has a built-in speaker and front-panel 4-pin locking headset jack, front- and
rear-panel mic inputs, analog I/O with XLR and RJ-45 connectors, GPIO speaker mute/ dim control. An
Ethernet jack completes the connection complement.
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Desktop Stations
IC.1D 20-Station Filmcap Intercom Panel
The IC.1D 20-station desktop intercom is perfect for producers, screeners, etc. IC.1D has 20 preset
stations presented on LED-backlit button caps; an economical way to add intercom function to any
space. 20 LED-backlit film-cap buttons can be labeled with station names and programmed using a
built-in Web interface and any browser. The OLED callback window lets users identify and answer calls
from remote stations that aren’t programmed on a local “speed” key. IC.1D includes a built-in speaker
and front-panel 4-pin locking headset jack. All it takes to add it to your intercom network is a single
CAT-5 connected to the rear-panel Ethernet port; a built-in auto-sensing power supply eliminates
nasty “wall warts.”
IC.20D 20-Station OLED Intercom Pane
The IC.20D is the desktop version of the IC.20 rack-mount station we showed you earlier. The 20 station
preset locations are equipped with high-resolution OLED displays; the OLED callback window and
dialing pad let operators call any station not programmed to a preset location. Naturally there’s a built-in
speaker, front-panel 4-pin locking headset jack, front-panel mic input, an Ethernet port for fast hookup,
and internal auto-sensing power supply.
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AXIA | IP INTERCOM
Console Modules
You don’t need to own an Axia console to use IP Intercom — rack-mount and desktop stations integrate
with any broadcast mixer to route intercom traffic to air instantly — full-bandwidth, broadcast-quality
audio, not tin-can-and-string noise. But if you do own an Axia Fusion or Element mixing console, these
drop-in modules make communications even easier by turning your board into an intercom station!
Built-in Talkback functions enable seamless communication between board ops, hosts and studio guests.
20 Station OLED Intercom Module
The 20-Station OLED intercom module requires two frame positions and provides
access to 20 pre-programmed intercom stations. Individual talk and listen buttons are
combined with high-resolution OLED displays for fast access to frequently-called stations;
auto-answer functions are also provided. Mic audio is taken directly from the console
operator’s microphone; speaker audio is directed to the console’s preview speaker.
There’s a dedicated listen volume control, individual mic and speaker mute keys and
group talk functions; the overbridge display works with the console’s monitor module
numeric keypad to give direct access to any station systemwide. Station presets and GPIO
functions are programmed using any standard Web browser. Available for Fusion and
Element consoles.
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10 Station OLED Intercom Module
The 10-Station OLED intercom module occupies one console frame position and includes ten
preset locations with 10-character OLED displays, auto-answer functions, dedicated listen
volume control, and mute keys for speaker and mic. Available for Axia Element console only.
10 Station Film-Cap Intercom Module
This economical 10-Station Film-Cap intercom module features ten LED-backlit film-cap
buttons for single-button calling of up to 10 preset stations. This module occupies one frame
position, and also provides a dedicated listen volume control, speaker and mic mute buttons. It
uses a single frame position. Available for Fusion and Element consoles.
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SoftCom IP Intercom for Windows
Axia Softcom Intercom for Windows makes any networked PC a part of your IP Intercom system! The
easy user interface mimics the IC-20 control panel, with preset locations for 20 frequently-called
stations. Auto-answer and hands-free functions are supported, and a drop-down station finder
gives instant access to stations not pre-programmed. All your PC needs is a sound card with mic &
speakers, and a 100BASE-T Ethernet connection to your Axia IP-Audio network. Purchase includes a
site license for all PCs.
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SPECIFICATIONS
Like all Axia products, IP Intercom uses only premium, studio-grade audio components to guarantee
maximum performance.
Microphone Preamplifiers
• Source Impedance: 150 ohms
• Input Impedance: 4 k ohms minimum, balanced
• Nominal Level Range: Adjustable, -75 dBu to -20 dBu
• Input Headroom: >20 dB above nominal input
• Output Level: +4 dBu, nominal
Analog Line Inputs
• Input Impedance: 20 k Ohms
• Nominal Level Range: Selectable, +4 dBu or -10dBv
• Input Headroom: 20 dB above nominal input
Analog Line Outputs
• Output Source Impedance: <50 ohms balanced
• Output Load Impedance: 600 ohms, minimum
• Nominal Output Level: +4 dBu
• Maximum Output Level: +24 dBu
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: 125 dB
Equivalent Input Noise
• Microphone Preamp: -128 dBu, 150 ohm source, reference -50 dBu input level
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Total Harmonic Distortion + Noise
• Mic Pre Input to Analog Line Output: <0.005%, 1 kHz, -38 dBu input, +18 dBu output
• 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 and CMRR
• Analog Line channel to channel isolation: 90 dB isolation minimum, 20 Hz to 20 kHz
• Microphone channel to channel isolation: 80 dB isolation minimum, 20 Hz to 20 kHz
• Analog Line Input CMRR: >60 dB, 20 Hz to 20 kHz
• Microphone Input CMRR: >55 dB, 20 Hz to 20 kHz
Power Supply AC Input, rackmount and desktop stations
• Auto-sensing supply, 90VAC to 240VAC, 50 Hz to 60 Hz, IEC receptacle, internal fuse
• Power consumption: 35 Watts or less
Operating Temperatures
• -10 degrees C to +40 degrees C, <90% humidity, no condensation
Dimensions
• IC.20: 3.5 inches x 19 inches x 8.5 inches, 5 pounds
• IC.10X: 1.75 inches x 19 inches x 8.5 inches, 4 pounds
• IC.10: 1.75 inches x 19 inches x 8.5 inches, 4 pounds
• IC.1: 1.75 inches x 19 inches x 8.5 inches, 4 pounds
• IC.20D: 18.25 inches x 6 inches x 5.75 inches, 6 pounds
• IC.1D: 13.5 inches x 8.5 inches x 4.5 inches, 6 pounds
SoftCom PC Hardware Requirements
• Windows XP or higher
• 20MB free hard drive space
• 100BASE-T Ethernet connection to Axia network
• Sound card and mic/earphone headset
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25-SEVEN SYSTEMS | VOLTAIR
Voltair®
Ratings. Confidence.
OVERVIEW
It’s all about your listeners. Your people, your programming, your gear… they’re all focused on building
and retaining your audience.
We don’t have to tell you about the direct link between the size and composition of your audience—as
measured and reported by your ratings—and your advertising revenue. That’s why it’s vital to you that
every panelist in your market is accurately measured and that every station is playing on a level field.
Introducing Voltair, designed to give you greater confidence that every listener is counted when it
counts the most.
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25-SEVEN SYSTEMS | VOLTAIR
Ratings, Programming & Technical Operations
Radio ratings have been called “a game of inches,” where winners and losers are sometimes decided by
the thinnest of margins. Station management teams have always carefully monitored their markets’
listener data and taken action to maximize ratings and revenue.
With changes in rating survey methodologies in recent years, many program directors report making
more dramatic changes than ever before. For example, dayparts have been moved, local breaks have
been reduced, and programming clocks have become more rigid in response to the hard, quarter-hour
boundaries of ratings credit. Likewise, the industry has seen changes in audio processing practices,
airchain device order and other station engineering procedures—all in service to optimal performance of
watermark-based ratings technologies.
Because ratings performance data is provided on a delayed basis, stations lack the means to conduct
real-time analysis of audience response. Programmers have had limited insight into what efforts are of
benefit and why. Hence, most station efforts to optimize their performance in ratings have been trial and
error, with little insight into what may or may not be effective.
Understanding The Current Ratings Ecosystem
25-Seven® has been following the deployment of watermark-based rating methodologies since they
were introduced. We’ve spoken with many program directors and engineers, getting their perspectives
on the overall system architecture and the results of their optimization efforts.
After researching the publicly available data on the technology, our team of broadcast and audio experts
uncovered the variables that contribute to watermark integrity. More importantly, we developed Voltair
to provide you with the tools you need to monitor and analyze these variables, providing you with data
to inform your technical and programming decisions.
What did we find?
• Audio Content—The spectral characteristics of your audio content—music, announcer voices, etc.—
may negatively impact the robustness of your watermark encoding. Simply put, some audio content
encodes well while other content does not.
• Listener Environment—A listener’s device may not detect particular content because of their current
acoustic environment. A song or voice may not, for example, decode as well in a car as it does in a
bedroom.
To address these issues, we needed to account for the highly complex set of interactions among the
encoding and decoding processes, the audio properties of content, and listeners’ acoustic environments.
We developed a totally new set of easy-to-use tools for Voltair to analyze and help you manage the
consequences of these interactions.
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25-SEVEN SYSTEMS | VOLTAIR
How Voltair Can Benefit You
Operating transparently in your airchain, Voltair:
• Monitors and analyzes the robustness of watermark encoding across all program content.
• Offers visibility into how listening environments may influence watermark decoding, using models of
acoustic spaces where listeners are wearing or carrying their devices.
• Includes advanced audio signal processing to enhance the detectability of the watermark codes within
the context of your programming objectives.
• Empowers programmers to make informed decisions to address potential weaknesses in either
encoding or decoding.
For example:
• You can compensate for changes in program material and listening environments during different
dayparts and program types.
• You may choose to balance strong and weak program segments within each quarter-hour time
segment to produce successful decoding and get the earned credit for the full segment.
Voltair also serves as an off-line tool to identify produced and live content with low encoding confidence.
New programming elements—liners, promotions, etc.—may be created with greater confidence of
strong encoding.
Voltair and Your Ratings
Your revenue relies on your ratings. The overriding purpose of Voltair is to increase your confidence that
those ratings accurately reflect the habits of your listeners.
As you work to get credit for all the ratings you’ve earned, keep the following in mind:
• Even one listener device worth of data can make a measurable difference in your ratings.
• Insights into how your audio content is handled during the watermarking process can show you ways
to improve the robustness of your encoding.
• Consideration of listening environments can guide changes that may improve the reliability of
watermark decoding on listener devices.
• When you have confidence in the end-to-end watermark system performance of your stations’
signals, you also have more confidence in the relationship between ratings and your programming
decisions.
Voltair won’t increase your actual listenership, but it will help you be more confident that listeners to
your station participating in the watermark-based ratings process are correctly measured.
Regulatory
North America: FCC and CE tested and compliant, power supply is UL approved.
Europe: Complies with the European Union Directive 2002/95/EC on the restriction of the use of certain
hazardous substances in electrical and electronic equipment (RoHS), as amended by Commission
Decisions 2005/618/EC, 2005/717/ EC, 2005/747/EC (RoHS Directive), and WEEE.
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25-SEVEN SYSTEMS | TVC-15
TVC-15
Broadcast Watermark Analyzer, Monitor, and
Adaptive Controller
OVERVIEW
TVC-15 Overview
Broadcasting is a numbers business. Your success depends on what kind of audience you attract and
hold. Audience size and composition is measured primarily by reports from private ratings agencies,
and for most broadcasters, there’s a direct link between those reports and a station’s revenue. In
electronically measured markets, having good tools—ones that help you understand the entire
electronic measurement ecosystem—is essential to your station’s competitive picture. With TVC-15,
for the first time ever, you can detect, monitor and analyze how well each element in your programming
supports watermarking. Measurements happen in real time, right off the air, without depending on or
being connected to a particular encoder. Every 400 milliseconds, TVC-15’s tone verification codec analyzes
the actual code symbols in any audio you feed it, whether yours, or your competitors’. It will work from
any source, live or recorded. A front panel graph of your station’s watermark density gives you a granular,
moment-by-moment display; you can also download reports to look at encoding quality over hours,
days and weeks. AUDIO TIME MANAGEMENT | WATERMARKING MONITORING & PROCESSING
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25-SEVEN SYSTEMS | TVC-15
And for stations with a Voltair watermark monitor and processor, you can use TVC-15 to automatically
adjust enhancement levels in real-time. TVC-15’s Intelligent Adaptive Enhancement [AE] closes the
feedback loop, letting you dynamically control Voltair processing based on moment-by-moment analysis
of your actual air signal, pushing enhancement when it is needed, while backing off when not. For more
advanced watermark monitoring, TVC-15 lets you See What Counts!
IN DEPTH
Electronic Measurement and Your Ratings
Broadcasting is a numbers business. Your success depends on what kind of audience you attract and
hold. Audience size and composition are measured primarily by reports from private ratings agencies,
and for most broadcasters, there’s a direct link between those reports and a station’s revenue. The
viability of your station’s watermarks is constantly varying, depending on your programming, the
panelists’ environments, and other variables. Changes can happen as quickly as individual syllables in an
announcer’s voice, or traffic noises on the highway.
How Watermarking Works
Ratings agencies bases these reports on listener panels, where each panelist represents many people
in a market. In electronically measured markets such as the top 48 markets in the USA, panelists
wear portable devices called meters. These meters register unique digital codes broadcast by each
cooperating station. Thousands of these codes can be created in the course of an hour. In theory,
whenever a panelist hears a station—on their car or home receivers, in a store or restaurant, or even
from a colleague’s Internet computer speaker—the meter hears the station’s code, and the ratings
system registers the listening.
The codes themselves sound something like a fax signal, and aren’t pleasant to the ear... so they’re
deliberately ‘masked’ under louder sounds in the programming, in a process called watermarking.
Masking is a psychoacoustic phenomenon that keeps us from hearing certain combinations of sounds,
even though electronic meters can still detect them. But there are more than a hundred possible digital
code symbols used by the meter-based system, and each requires slightly different characteristics in
the masking sound.
A proprietary watermarking encoder provided by the ratings agency sits in your air chain, and looks for
masking opportunities where it can embed hidden codes. When it hears a potential mask for a current
digital code symbol, it generates the symbol and mixes it with the programming. Unfortunately, masks
are evanescent, appearing and disappearing as your content changes... sometimes, many times per
second. So the number of codes you can broadcast is also constantly changing, depending on your
programming. Some content is a lot better at supporting watermarks than others. Silence doesn’t
support them at all.
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25-SEVEN SYSTEMS | TVC-15
• Got a talk show with a musical introduction? Chances are the intro will have more encoding opportunities
than the talk.
• Running a sports show or drama? Scenes with just play-by-play or dialog probably won’t be encoded as
well as those with crowds or other busy backgrounds.
• Playing a commercial or promo? Our research indicates a sung jingle usually encodes better than a dry
voice-over... even though the spoken words might be more important to the selling message.
Furthermore, masking requires the code symbol to be significantly softer than the masking audio.
As your content gets softer, the encoding hardware has to make the codes softer. Environmental
noise around the listener can interfere with those softer codes, even if your listeners don’t mind the
noise: Humans are very good at tracking meaningful voice or music in a noisy environment. Meters,
unfortunately, aren’t as smart: It’s possible that a watermark signal, sent by the encoder at levels where
it wouldn’t be annoying in a quiet environment, doesn’t get detected by panelists’ meters in the real,
noisy world.
Bottom Line
The viability of your station’s watermarks is constantly varying, depending on your programming, the
panelists’ environments, and other variables. Changes can happen as quickly as individual syllables in an
announcer’s voice, or traffic noises on the highway.
Having good tools—ones that help you understand the entire electronic measurement ecosystem—is
essential to your station’s competitive picture.
What can be done?
25-Seven put years of research and testing into the technical issues with watermarking, and our
groundbreaking Voltair processor works with your station’s encoder to enhance watermarking codes as
they’re being generated. Voltair’s enhancement can be varied
by the station to accommodate different
programming styles, and controlled by station automation for different dayparts.
Many stations have found Voltair effective to help make their electronically derived ratings a better
match for the audiences they know they’ve got, and more reliable during hard-to-encode programming.
But to really manage this kind of problem, you have to be able to quantify it.
Both Voltair and hardware provided by ratings agencies include ways to meawsure how encodable a
program stream is. Voltair can be particularly helpful, with a minute-by-minute front panel display
of code reliability, techniques to reduce randomness when calibrating Enhancement settings against
ratings reports, and optional downloadable history reports and Excel graphs of a station’s coding activity.
But neither system can give you moment-by-moment measurements of how well each element in your
programming supports watermarks.
And neither system takes this information to the next level, actually adjusting enhancement levels in realtime to compensate for the wide variety of sounds that keep a radio station interesting.
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25-SEVEN SYSTEMS | TVC-15
You need to understand the entire electronic rating system. You need tools that can quickly and precisely
measure how it works. And you need efficient ways to apply this knowledge so it can optimize your
station’s product.
That’s why we developed TVC-15.
SPECIFICATIONS
We are constantly working to improve our products. Specifications and features are subject to change
without notice
ANALOG LINE INPUTS:
• Input Impedance: >40 k ohms, balanced
• Nominal Input Range: +4 dBu
• Input Headroom: 20 dB above nominal input
• A/D Conversions: 24-bit, Delta-Sigma, 256x oversampling
POWER SUPPLY AC INPUT
• Auto-ranging supply, 100VAC to 240VAC, 50 Hz to 60 Hz
• IEC receptacle, internal fuse, on/o switch
• Power consumption: 55 Wa s
OPERATING TEMPERATURES
• 0 degree C to +40 degree C, <90% humidity, no condensation
DIMENSIONS AND WEIGHT
• Chassis Dimensions (ex protrusions): 19” (48.2 cm) wide
3.5” (8.9 cm) height
11.75” (30 cm) depth
• Chassis Weight: 14.5 lbs. (6.57 kg)
• Shipping Dimensions & Weight: Contact customer support
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25-SEVEN SYSTEMS | TVC-15
FEATURES & BENEFITS
Every 400 milliseconds — 150 times per minute — TVC-15’s tone verification codec analyzes the
actual code symbols in any audio you feed it.
• Raw symbol reliability is displayed on a constantly changing bar. The symbols that make up a complete
station identification message are then processed through our proprietary algorithms.
A front panel graph of your station’s watermark reliability updates every 400 milliseconds.
• That’s fast enough to track individual program elements, or style changes in a song, or even the
difference between a host and a call-in guest.
A front panel timer updates every time your station broadcasts a complete watermark message.
• It takes 4.8 seconds for the watermark system to assemble enough code symbols for full station
identification. Under ideal circumstances1, TVC decodes a complete message every 4.8 seconds. Each
time you do, the timer resets and appropriate message details are displayed.
• During periods of low masking (silence, spoken word, some music), the timer doesn’t get as many
chances to reset. It keeps counting, and changes color to alert you to the condition.
TVC-15 doesn’t depend on a particular encoder, and doesn’t have to be connected to it.
• You can connect TVC to an air monitor. Or to an Internet radio, a HD receiver, or any other way listeners
are getting a signal with watermarking codes. Use any convenient analog source, and get an instant
reading of how strong its codes are.
• TVC is switchable between encoding formats: Layer 1 (used for US radio) and Layer 2 (Canada and
some other countries).
• You can equalize or distort the signal going to TVC to simulate low-quality radios. Or you can feed TVC
from a microphone pointed to any radio or loudspeaker, in a quiet test room or noisy public space2.
• You can bias TVC’s measurements using statistical noise simulation. Or you can record actual
environmental noise and other possible interference, and mix it with the signal you’re feeding TVC.
• You can feed it other stations’ signals, to assure code reliability across a broadcast group... or even see
how your competition is encoding. All this can happen in the privacy of your own local network, with
nobody else able to see how you’re making programming decisions.
• TVC’s front panel and reports even identify when it sees different encoders, so you can scan multiple
signal sources and sort them out later.
TVC-15 will work from any source, real-time or recorded.
• You can feed TVC recordings of your own or other stations’ signals, whether they’re from your program
line, off-air monitors, or recordings from public spaces.
• You can use it offline with a spare encoder, to analyze program segments or production elements.
TVC’s fast response lets you compare different sub-elements within a program stream.
• You can use it with an automated switcher to cycle among various stations and program streams in
your group to verify that encoders are working.
• Operation is completely flexible: Input can be switched between program sources or among different
encoders without the need to recalibrate or reboot.
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25-SEVEN SYSTEMS | TVC-15
TVC-15 gives you downloadable reports and remote readouts.
• You can access TVC’s password protected real-time display from any connected computer, even over
the web. You’ll know in an instant how well your programming is supporting watermark codes.
• You can download csv-formatted daily history reports of minute-by-minute actual code reliability, for
custom analysis or for display in a program like Excel. Reports are private and you control who sees
them.
And optionally, the big benefit for Voltair users:
TVC-15 can control your Voltair in real-time!
TVC-15’s Intelligent Adaptive Enhancement [AE] closes the feedback loop, letting you dynamically
control Voltair processing based on moment-by-moment analysis of your actual air signal.
You can take coding enhancement beyond simplistic “set and forget” or daypart setting strategies.
TVC and Voltair work together like a continuous, intelligent automatic gain control on your hidden
watermarks!
Have male and female hosts in a conversation? Got a call-in guest on a very compressed cell phone?
Airing a stopset with jingles, dry announce, and produced sweepers? TVC-15 lets you compensate for all
their different encoding requirements, continuously and with minimum annoyance to your listeners.
• Feed TVC-15 with your air signal, give it your Voltair’s log-in address, and AE will constantly adjust
your connected Voltair to provide just enough enhancement for the watermark confidence you want
to achieve...while protecting the sound you want for your station, with minimum noticeable processing
changes and artifacts.
Benefiting From TVC-15
Monitoring & Analysis of Station Encoding
It’s vital to know that your watermarking system is working properly. Common wisdom in radio today is,
“If you aren’t encoding, you might as well be off the air.”
But there aren’t many ways to verify when you’re encoding. The standard watermark encoder provides
only a simple “no code” warning and basic error messages on an LCD. It won’t alert you if a program
stream is only marginally supporting watermarks. You might miss a lot of message opportunities before
there’s an alert.
The monitoring facility in Voltair is more powerful, sending initial warnings when 15 seconds have gone
by without a valid message, and adding more warnings as the condition gets longer. Its front panel
and optional downloadable reports give a minute-by-minute analysis of coding confidence, and let you
simulate how various forms of environmental noise will affect it5.
But both the standard encoder and Voltair’s analysis can look only at the codes as they are being
generated. Before those codes get to a listener, they’ll often pass through a composite clipper or some
data compression. Then they can be hit with transmitter issues or RF interference. In some installations,
watermarking is also affected by airchain equalization or multiband compression.
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25-SEVEN SYSTEMS | TVC-15
You wouldn’t consider your audio monitoring complete without a tuner, internet receiver, or some
other form of real-world verification.
TVC-15 lets you do the same thing for your encoding.
You can feed TVC-15 with any audio signal, from a monitor receiver, a consumer radio that’s flipping
station-to-station, a field recording, a remote microphone, a router or patchbay... any source of
analog audio.
On top of that, our sophisticated algorithms bring confidence analysis to levels that were never before
possible with any system.
Near-instantaneous response:
• TVC-15’s signal strength bar continuously responds to signal strength in the frequencies used by
watermarking.
• It takes 400 ms for the encoder to create a valid code symbol, so TVC’s front-panel graph updates
that quickly: 150 times per minute. That’s fast enough to indicate the differences when two on-air
hosts have a conversation, or distinguish a sung jingle from a donut voice-over. The most recent two
minutes of confidence measurements are displayed on a scrolling graph.
• A complete identification message requires 12 valid code symbols, carried on a combination of 10
different frequency channels. As soon as a valid ID is received, TVC’s front-panel timer starts counting.
If it takes too long for TVC to see a new valid message, the timer changes color.
More detailed information:
• A detailed 0 – 100% display of the likelihood each potential message will be received.
• Identification tags for each encoder. You can tell at a glance which of your streams—or your
competitors’—is being analyzed.
• The timestamp encoded in each successful message. You can tell at a glance if an encoder’s clock isn’t
accurate, a situation which can interfere with reliable ratings.
Complete remote access:
• TVC has a built-in, password-protected web server. You can log in with any connected browser, and
assign different users the ability to either monitor TVC’s readings, or remotely control its behavior.
Downloadable full reports:
• TVC’s internal web server also lets you download a complete analysis of every signal TVC has received,
available for any day it’s been turned on. TVC reports are available as detailed files of each 4.8-second
complete message analyzed over the course of a day, or as one-minute averages. They’re in csv
format, so you can analyze them with your own software, display them as an Excel spreadsheet, or
compare them with station ratings reports. Reports are available only by password-protected log in:
You control who sees the data.
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25-SEVEN SYSTEMS | TVC-15
Controlling Voltair Enhancement in Real-time
Voltair caused a revolution in station processing, enhancing watermarks so they’d have a better chance
of being picked up by panelists’ meters... even when a signal didn’t support watermarking perfectly, or
when a panelist was in a noisy environment. Voltair doesn’t create ‘phantom panelists’ in the ratings
system, but it helps make sure stations get credit for the listeners they really have. Unfortunately, too
much enhancement can actually discourage listeners, breaking through the masking phenomenon,
making watermark messages audible in the program stream. Listeners may hear this as extra noise or
distortion. In extreme situations, they can be chased away.
It’s a question of balance: You need enough enhancement to make codes reliable even during hard-toencode program segments, or when there’s a lot of environmental noise. But you don’t want to annoy
listeners. How much enhancement is too much? It depends on the program material, listening situation,
and even listener expectations—the right enhancement for a news talk show might be too much for a
high-quality acoustic music set.
Voltair includes tools including a “toggle test,” to calibrate the amount of enhancement. It lets you add
controlled amounts that can be correlated with ratings reports, so users can run their own tests. It also
lets you preset three different Enhancement levels with GPIO control: You can have an “emergency
watermark boost” button in master control, change enhancement when the host turns on his
microphone, or have your station’s automation system change enhancement for different dayparts.
But to get the highest level of control you’d need a trained operator, constantly monitoring your actual
on-air signal with TVC, and continuously adjusting Voltair’s enhancement for different air talents, audio
sources, noise levels, and quality requirements. An operator who knows the personality and sound you
want to present. One who’s subtle enough to control watermark enhancement while avoiding abrupt or
annoying changes. One who can pay perfect attention 24 hours a day, 7 days a week...
TVC-15’s Intelligent Adaptive Enhancement can be that operator.
TVC-15, together with Voltair, closes the feedback loop around your watermarking ecosystem. It acts
as a “smart AGC” for Voltair enhancement, monitoring actual encoding, and adjusting the amount of
enhancement as quickly as twice per second. But like a good transmitter processor, you can fine-tune its
behavior to preserve your station’s unique sound, setting minimum desirable confidence levels, as well
as maximum enhancement to annoying artifacts, how quickly enhancement can be changed, and more.
Finally: complete, full-time control over ratings enhancement levels!
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25-SEVEN SYSTEMS | TVC-15
FRONT PANEL
TVC-15 gives you a live, highly detailed display of actual watermark symbols, evaluated every 400 ms.
for confidence, completeness and reliability.
Our proprietary algorithms constantly analyze the input signal, looking for valid code symbols that the
system combines to build meaningful station identifications. The input can be any real-world source:
your off-air signal (or a competitor’s), a test file from a production studio, an Internet stream, or even a
live mic listening to a sample radio or a public space7. If there are symbols hidden in the audio, TVC-15
will report their details.
The front panel LCD is arranged for maximum usability:
Time since last complete message
This reports minutes and seconds since the last successfully decoded message. It flashes green and
restarts from 00:00 whenever a complete and coherent message is received.
• Continuously short timings are good: They mean the program includes a lot of reliable messages. The
display will be green.
• Longer times between restarts mean the programming isn’t supporting codes well. The display
changes to yellow if ten seconds have gone by with no messages, to red if thirty seconds have gone
by. While there may be exceptions, chances are a station won’t be identified during those times.
• The Interval Display is constantly updating, and gives you a quick go/no-go indication of the
current signal.
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25-SEVEN SYSTEMS | TVC-15
Last Complete Message Received
This is based on the actual Encoder ID that accompanied the last valid message, along with an optional
display of the time stamp that accompanied it. Encoder IDs are arbitrary and set by the ratings agency,
and don’t include a station’s call letters or frequency. So TVC-15 identifies them simply as Encoder A,
Encoder B, and so on. You can rename them easily (to show call letters, frequency, HD stream, or any
other useful tag), and TVC will use that name every subsequent time it sees that encoder.
The end of this line includes a short nickname in quotes. This nickname is used for flags at the bottom of
the Main Confidence Graph.
Simulated Environmental “Noise Loading”
If everyone listened to broadcasts using headphones, the signal would go straight from the receiver
into human ears. If they also used an adapter cable, it could go straight into a panelist’s portable meter
as well. But most listening is done with speakers, and in a variety of acoustic environments. Whether
a panelist is driving their car, attending a sports event, or in a bar that has radio or TV for background,
ambient noise is a factor that can affect how portable meters receive your code. So, to help gauge the
impact of different noisy environments, we let you apply various levels of simulated noise.
25-Seven’s Voltair is designed for real-world, real-time watermark evaluations while a station is
broadcasting. It lets you simulate different listening situations with built-in recordings of actual randomnoise environments (traffic with car honks, households with baby cries, dishes clattering in restaurant)
and apply them to your measurements.
TVC’s, however, can also be used for accurate offline comparisons of different program streams. The
randomness of real-world noise can affect these comparisons, depending on each programs’ timing.
So TVC can generate a signal to simulate real-world noise in a repeatable way. It acts as a constant
“load” on the watermark energy. It lets you compare different programs with the confidence that
environmental noise will have a similar influence on each. You can also use this Noise Loading to scale
TVC’s measurements, for more convenient analysis and graphing.
If you want, you can substitute your own noise source instead. This can be recorded environmental
noise that you mix with the test signal before feeding to TVC. Or it can be a live mic in a real-world space,
picking up both your program and the location’s actual noise.
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25-SEVEN SYSTEMS | TVC-15
This shows the confidence level for complete messages during the past two minutes. A complete
message consists of twelve individual code symbols in a valid pattern, so TVC-15 draws a new
Confidence line every 400ms. The line height displays zero to 100% confidence, and its color provides a
quick visual reference:
• Dark green lines indicate 80% confidence or better. This can be the result of programming choices,
Voltair enhancement, or a combination of both.
• Light green lines show at least 40% confidence. Many of your watermarks will probably get through,
unless there’s a lot of environmental noise.
• Orange lines show at least 30% confidence. Watermarks may be getting lost. Red lines show less
than 30% confidence. There’s a good chance panelists’ meters won’t register your station at all, even if
they’re actively listening.
• No line at all is rare, but can occur during prolonged silences.
Code Symbol Strength Bar
This white line constantly changes height to show the strength of potential code signals in watermark
channels. This bar reacts instantly, to provide visual feedback that encoding could be taking place. Actual
code symbols require 400 ms to broadcast, and they’re measured and displayed in the Main Confidence
Graph.
Encoder Nickname Tags
These are abbreviated from the encoder name, and mark each time a complete message comes through.
The tag will be either the last word of the name, if it’s short enough to fit on the graph; or the first three
letters of the last word.
Two Minute History
The time display on the bottom of the Confidence Graph is calibrated in minutes: seconds, based on
TVC’s real-time clock10, to help you correlate confidence readings with moment-by-moment changes in
your programming. This is not the time-stamp encoded on watermark messages.
Other User Controls
TVC-15 includes complete, flexible control over its operation. Clock and system settings, network access,
how TVC controls a connected Voltair, and remote passwords can be set from the front panel. Most of
these settings, along with maintenance and customization functions, are also available remotely using
any web browser.
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25-SEVEN SYSTEMS | PROGRAM DELAY MANAGER
Program Delay Manager
Profanity Delay Reinvented
OVERVIEW
• PD Alert™ instantly emails time-stamped audio files whenever Dump is pressed
• Files capture what took place both on- and off-air
• Seamlessly builds and exits delay
• Configurable delay time, build and dump options
• Delayed IP data, serial streams and GPIO sync’d to audio
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25-SEVEN SYSTEMS | PROGRAM DELAY MANAGER
FEATURES
It’s About Time
Leave it to 25-Seven® Systems to re-invent the profanity delay. Program Delay Manager (PDM) brings
the possibilities of the Internet age to a “stand-alone box” technology that hasn’t advanced much since
the 1980’s. Ease of use, transparent audio quality and program director friendly features converge in
PDM to take an old process to a new level.
The Air Check is in the Email
Program Directors have more on their plates today than ever before. There’s no way anyone can monitor
every broadcast hour of every day, but PDs need to be the first to know what happened when that
“dump” button got pressed.
With Program Delay Manager’s patented PD-Alert™ feature, two time-stamped audio files capturing
what took place both on air and off air get internally archived and emailed to the PD (or GM, or CE, or the
legal team) every time questionable material is “dumped”.
For stations serious about protecting their license, PDM provides an instant log record establishing your
station’s action and intent to keep the airwaves clean.
99 Seconds Of Delay Your Way
PDM comes standard with 99 seconds of stereo audio delay, and a dump button that can be set to
remove any number of seconds you choose.
Build a delay through pre-rolling, time expansion or audio file play-out capabilities built right into PDM.
Exit a delay through time compression or use the Cough button to simply wait and exit.
Dump audio through the standard “cut and rebuild” method, or use PDM’s Overkill™ feature to play a
“fill” file. Overkill allows you to select a show specific file from a list and play it over the dump buffer
instead of collapsing the delay.
How PDM Does It
Superior Audio Algorithm Quality
25-Seven has a well-deserved reputation for offering the industry’s most transparent time compression
and expansion algorithms. Your listeners probably won’t appreciate our superior, artifact-free audio
because they won’t perceive it’s in use!
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25-SEVEN SYSTEMS | PROGRAM DELAY MANAGER
Flawless Expansion/Compression
25-Seven Systems’ imperceptible audio time compression algorithms serve up smooth, crisp, stutterfree audio in PDM, even on stereo music. Unlike other products, we never splice at level thresholds or
alter pitch. Clean audio is what we do best… now you can be sure the content is “clean” as well! Better
algorithms mean delays can be rebuilt faster, so you can safely get back to callers. Build or Exit rates can
be adjusted in real time, so you can be more or less aggressive, depending on audio content.
Audio, RDS, Data Streams and GPI/O Stay Synced
PAD or “now playing” data streams are delayed in precise synchronization with the audio as it grows,
shrinks or whenever the dump button is pressed. PDM’s data-follow-audio capabilities allow flexible
synchronization from any data input to any data output. For example, serial data entering the RS-232
input can be routed to an IP output while remaining synchronized to the audio. 2 independent data
delays are supported, and GPI/O closures stay in sync, too.
Future-Proof Audio Quality
Superior balanced analog I/O, with AES digital standard. 85dB s/n, response 25Hz-18kHz (+0/-0.2dB)
and 0.02% THD+N… even during compression/expansion. Audio is always linear, so no lossy data
reduction enters your signal path.
AES Digital, Balanced Analog or Livewire® AoIP
The first program delay to provide Audio over IP (AoIP) and control over Axia® Livewire audio networks,
PDM comes in two models: one with balanced analog and AES digital I/O and the other with AoIP for
Livewire. Whether you already have a Livewire network or you want to keep your plant AoIP-capable,
PDM has you covered with Ethernet connectivity.
Superior Control
Choices, choices! PDM presents you with easy-to-use front panel controls, designed for the rigors of
radio. Contact closure commands can be synced to the audio delay by the smart, programmable 8x8
GP I/O. Full bi-directional serial control over both RS232 and IP include advanced real-time status
monitoring of parameters such as current delay depth and audio levels. A comprehensive web interface
allows your PDM to be managed from nearly anywhere. Our Multi-View web feature permits networks
and big facilities to monitor and manage up to 20 PDM’s from a single browser screen.
Web Configurable
Say goodbye to hieroglyphs. Navigating through “set and forget” parameters is a breeze with our built-in
web server. Change your settings, upload audio files and manage PDM’s dump archives using simple
browser screens, so you don’t waste time trying to enter data though an ill suited LCD interface. Talk to
PDM over your LAN or WAN. What could be easier?
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25-SEVEN SYSTEMS | PROGRAM DELAY MANAGER
IN DEPTH
Web Interface
In addition to controlling Program Delay Manager using GPIO (contract closures), PDM comes with a
built-in, password-protected web server, allowing you to remote control your unit across a local or wide
area network.
The server gives you five separate pages for complete and convenient control over your PDM.
Front Panel
An Adobe Flash-based application replicates PDM’s front panel on your web browser, so every
button and display is present and functions just like the real front panel. Through careful client-server
communications management, round-trip latency is almost imperceptible, creating a seamless user
experience. You can even control PDM from multiple computers. Just open a web browser interface on
each, and anything you do on one computer will be reflected on the others, as well as on PDM’s physical
front panel.
Configuration
Tired of learning hieroglyphics just to configure a profanity delay? Navigating though “set and forget”
parameters is a breeze with the PDM’s Configuration page. You’ll find obvious control with all your
settings on one simple screen, so you don’t waste time entering data though an ill-suited LCD interface.
PD Alerts™
A dedicated page lists all of the PD Alert emails the PDM has sent to your chosen staff.
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25-SEVEN SYSTEMS | PROGRAM DELAY MANAGER
Dump Archive
A Dump Archive shows all your saved dumped audio files. Easily review just what’s been cut out of your
air stream.
Insert Files
Easy management of the files you can use for quickly building your buffer at the beginning of the show.
No more flash drives and cryptic file names!
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25-SEVEN SYSTEMS | PROGRAM DELAY MANAGER
SPECIFICATIONS
Power Input
PDM comes with a standard IEC C14 power connector.
Network
PDM connects to a standard 100BASE-T network connection. This port is used for Axia Livewire,
synchronization to a network time server, and secure remote control via a web browser. If connected to
the Internet, it should be behind a hardware firewall.
Analog Inputs and Outputs
Stereo inputs are electronically balanced XLR females, pin 2 hot, with a load of 20kΩ: this makes it
compatible with all modern electronically-balanced outputs. If fed from a transformer-balanced output,
we recommend bridging a 680Ω resistor between pins 2 and 3. Outputs are electronically balanced XLR
males, pin 2 hot, designed to feed a load of 600Ω or greater. Input and output sensitivity default levels
can be set from the front panel, and can range between +20dBu and -10dBu.
Digital Inputs and Outputs
When set to AES/EBU via the configuration menu, this input conforms to IEC 958 Professional (5v p-p,
110Ω balanced) on XLR connectors. When set to s/pdif, the voltage and impedance switches to IEC 958
Consumer (.5v p-p, 75Ω unbalanced): connect signal to pin 2 and shield to pins 1 and 3. Digital output
(selectable AES or s/pdif) is always active, regardless of whether you are using analog or digital inputs.
PDM will lock to any valid 32 kHz, 44.1 kHz, or 48 kHz signal at the digital input connector, even if you
have selected analog for the input. In that case, the digital input controls PDM’s internal sample rate. If
PDM is not connected to a digital input, it uses its own high-reliability 44.1 kHz sample clock
Axia Livewire Version Inputs and Outputs
On the Livewire version of PDM, audio connections are exclusively via the network. PDM-Axia also
supports Livewire-based GPIO.
GPIO
Eight parallel control inputs and eight parallel control outputs appear on a DB-25 connector. Input and
output functions are assigned through a configuration menu on the front panel. Inputs and outputs are
opto-isolated for easy interface to other equipment. A +5v supply and ground are also brought out to the
DB-25 for simple remote controls using pushbuttons and LED status readouts. The +5v supply can carry
200 mA, more than adequate for 8 LEDs and 8 logic inputs. It is protected by an internal, self-resetting
thermal circuit breaker.
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25-SEVEN SYSTEMS | PROGRAM DELAY MANAGER
Detailed Specifications
Audio
• S/N ≥ 84 dBA with 10 dB headroom (≥94 dB dynamic range); THD @1 kHz < .01%; IMD (IHF) < .01%;
Frequency response ± 0.5 dB, 20 Hz – 20 kHz, measured analog input to analog output.
Dimensions
• 1RU (rack unit); 19” W (with rack ears) x 12” D x 1.75” H (483 x 305 x 44mm)
Power
• 100-240 VAC, 50/60 Hz; typical consumption 32 VA.
Regulatory
North America: FCC and CE tested and compliant, power supply is UL approved.
Europe: Complies with the European Union Directive 2002/95/EC on the restriction of the use of certain
hazardous substances in electrical and electronic equipment (RoHS), as amended by Commission
Decisions 2005/618/EC, 2005/717/ EC, 2005/747/EC (RoHS Directive), and WEEE.
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25-SEVEN SYSTEMS | PRECISION DELAY
Precision Delay
Your Station… In Sync and On Time
OVERVIEW
Saving Time, Managing Time
For nearly a decade, 25-Seven® Systems has been helping you solve your station’s time management
problems. Now we’ve got something for your toughest challenges. Precision Delay, our fourth specialized
product, addresses applications such as drift between analog and HD Radio transmission signals and
broadcast repeater synchronization.
Precision Delay offers sample-accurate delay times adjustable in fractions of a second; seamless
watermark-protecting builds and exits; synchronized data streams; network accessible control; and
25-Seven quality and support.
Keeping HD Radio in Sync with Analog
With more and more vehicles equipped with HD Radio receivers, stations can’t afford confusing listener
experiences due to blending out-of-sync analog and HD Radio signals.
Precision Delay lets you precisely set offset measurements by querying and retrieving them over IP from
your BELAR FMHD1 or AUDEMAT Golden Eagle modulation monitor.
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25-SEVEN SYSTEMS | PRECISION DELAY
Take Precision Delay Out to the Ballgame
When sports fans listen to radio play-by-play at the stadium, they may not know if they are in the right
ballpark when HD Radio diversity delay is running. Getting your station into “ball-game mode” means
switching the HD Radio signal on and off without annoying listeners or impacting ratings. Precision
Delay lets you smoothly build in and out of delay.
Watermark Friendly
Protecting the integrity of ratings watermark codes during delay builds and exits presents special
challenges. Precision Delay’s unique Watermark Safe Mode helps accommodate the time-based
structure of watermark encoding. Our algorithms never alter pitch, so unlike other time manipulation
processes, they never undermine the critical frequencies upon which watermarking depends.
Small Delays: Keep Boosters In Sync
Proper time alignment is critical to keeping main signals in sync with boosters or other transmitters
relaying on the same frequency. Precision Delay lets you adjust delays by increments as small as a single
sample.
Large Delays: Shift Across Time Zones
For facilities that need to delay content by several minutes to as much as four hours, Precision Delay
provides a flexible solution with no spinning hard drives and no complicated programming. With “set and
run” simplicity and solid-state reliability.
Delay Data & GPIO
Precision delay supports up to 3 independent data delays. Serial data over IP or RS-232 such as “now
playing” metadata can be delayed in sync with audio — even when delay time is in transition. Contact
closures can also be delayed on input so that they trigger against the appropriate audio on output.
AES, Balanced Analog and Livewire® IP Audio
AES digital, balanced analog and Livewire IP audio are standard. Whether you already have an IP audio
system or you want to keep your plant IP-capable, Precision Delay matches your signal path today and
for years to come.
Control Precision Delay from Anywhere
In addition to controlling Precision Delay using GPIO (contract closures), Precision Delay offers complete
configuration and control over a LAN or WAN using a common web browser. Navigating though
parameters is a breeze with our internal password-protected web server. The server gives you five
separate pages for complete and convenient control over your PD. The network interface also lets you
remotely install software updates. Whether your unit is located in your main equipment room or at the
transmitter, control is probably right where you’re sitting now.
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25-SEVEN SYSTEMS | PRECISION DELAY
Front Panel
An Adobe Flash-based application replicates PD’s front panel on your web browser, so every button
and display is present and functions just like the real front panel. Through careful client-server
communications management, round-trip latency is almost imperceptible, creating a seamless user
experience. You can even control PD from multiple computers. Just open a web browser interface on
each, and anything you do on one computer will be reflected on the others, as well as on PD’s physical
front panel.
Configuration
Tired of learning hieroglyphics just to configure a delay? Navigating though “set and forget” parameters
is a breeze with the PD’s Configuration page. You’ll find obvious control with all your settings on one
simple screen, so you don’t waste time entering data though an ill-suited LCD interface.
SPECIFICATIONS
Power Input
Precision Delay comes with a standard IEC C14 power connector.
Network
Precision Delay connects to a standard 100BASE-T network connection. This port is used for Axia®
Livewire, synchronization to a network time server, and secure remote control via a web browser. If
connected to the Internet, it should be behind a hardware firewall.
Analog Inputs and Outputs
Stereo inputs are electronically balanced XLR females, pin 2 hot, with a load of 20kΩ: this makes it
compatible with all modern electronically-balanced outputs. If fed from a transformer-balanced output,
we recommend bridging a 680Ω resistor between pins 2 and 3. Outputs are electronically balanced XLR
males, pin 2 hot, designed to feed a load of 600Ω or greater. Input and output sensitivity default levels
can be set from the front panel, and can range between +20dBu and -10dBu.
Digital Inputs and Outputs
When set to AES/EBU via the configuration menu, this input conforms to IEC 958 Professional (5v
p-p, 110Ω balanced) on XLR connectors. When set to s/pdif, the voltage and impedance switches to
IEC 958 Consumer (.5v p-p, 75Ω unbalanced): connect signal to pin 2 and shield to pins 1 and 3. Digital
output (selectable AES or s/pdif) is always active, regardless of whether you are using analog or digital
inputs. PD will lock to any valid 32 kHz, 44.1 kHz, or 48 kHz signal at the digital input connector, even if
you have selected analog for the input. In that case, the digital input controls Precision Delay s internal
sample rate. If Precision Delay is not connected to a digital input, it uses its own high-reliability 44.1
kHz sample clock.
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25-SEVEN SYSTEMS | PRECISION DELAY
Axia Livewire Inputs and Outputs
Precision Delay supports Livewire, with audio connections exclusively via the network. Precision Delay
also supports Livewire-based GPIO.
GPIO
Eight parallel control inputs and eight parallel control outputs appear on a DB-25 connector. Input and
output functions are assigned through a configuration menu on the front panel. Inputs and outputs are
opto-isolated for easy interface to other equipment. A +5v supply and ground are also brought out to the
DB-25 for simple remote controls using pushbuttons and LED status readouts. The +5v supply can carry
200 mA, more than adequate for 8 LEDs and 8 logic inputs. It is protected by an internal, self-resetting
thermal circuit breaker.
Detailed Specifications
Audio
• S/N ≥ 84 dBA with 10 dB headroom (≥94 dB dynamic range); THD @1 kHz < .01%; IMD (IHF) < .01%;
Frequency response ± 0.5 dB, 20 Hz – 20 kHz, measured analog input to analog output.
Dimensions
• 1RU (rack unit); 19” W (with rack ears) x 12” D x 1.75” H (483 x 305 x 44mm)
Delay range
• 10 ms to 4 hours; adjustable in 10 µs increments
Power
• 100-240 VAC, 50/60 Hz; typical consumption 32 VA.
Regulatory
North America: FCC and CE tested and compliant, power supply is UL approved.
Europe: Complies with the European Union Directive 2002/95/EC on the restriction of the use of
certain hazardous substances in electrical and electronic equipment (RoHS), as amended by Commission
Decisions 2005/618/EC, 2005/717/ EC, 2005/747/EC (RoHS Directive), and WEEE.
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Z/IPSTREAM X/2
Z/IPStream X/2
The future is streaming. The future is here.
OVERVIEW
WiFi and Internet connections are available everywhere these days — and so is streaming audio.
Already, many are using their smartphone as a 21st-century transistor radio. Soon, the connected car will
make it easier than ever to listen to high-quality streamed audio on the move. These are big changes in
listening habits, but don’t worry: Z/IPStream X/2 is here to help.
Z/IPStream X/2 is the third-generation streaming software from the Telos Alliance®; a new combined
audio processing/streaming platform designed for broadcasters who understand that streaming audio
quality and reliability are just as important as terrestrial transmission. Z/IPStream X/2 gives you the power to
fine-tune your streams for clear, clean, audio output — no matter the bitrate, codec, or delivery platform.
Z/IPStream X/2 stands above the rest with Adaptive Streaming technology. With Adaptive Streaming,
the connection between streaming server and listener is automatically managed, dynamically adjusting
bitrate and audio quality to maintain a solid connection with the best possible audio — regardless of
Wi-Fi limitations or Internet behavior. Z/IPStream X/2 supports generating multiple streams to a server
simultaneously using different codecs and bitrates to support these adaptive streaming applications.
PROCESSING + ENCODING FOR STREAMING AUDIO
TELOSALLIANCE.COM
Z/IPSTREAM X/2
FEATURES
• Genuine, high-quality audio codecs from Fraunhofer IIS (the inventors of MP3), including MP3, AAC-LC,
HE-AAC v1, HE-AAC v2, and xHE-AAC
• xHE-AAC works well at low bitrates and therefore has more encoding power. Whereas, other codecs
like AAC and MP3 sound much better for music than they do for speech, xHE-AAC sounds great for
both speech and music, even at the lowest bitrates
• Processes and encodes streaming audio for multiple platforms and bitrates simultaneously
• Includes 3-band processing from Omnia Audio
• Need even more processing power? Upgrade to Z/IPStream 9X/2, with up to seven bands of
multiband AGC and limiting plus Undo technology that can restore poorly mastered audio to clarity and
brilliance
• Sophisticated software routines enable you to handle streaming complications such as programming
blackouts, metadata insertion, variable listener environments, and more
• Flexible audio routing accepts input from sound cards, RTP and Livewire AoIP connections
• Unprecedented level of control: Use the built-in HTML5 web interface, or fine-tune even further using
the REST API
• Cloud-Ready: Z/IPStream X/2 may be hosted and run using your cloud-based server
• Built-in SNMP and email notification of system events
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Z/IPSTREAM X/2
IN DEPTH
Adaptive Streaming
Adaptive Streaming is a stream-delivery method that allows media players to switch bitrates when
network conditions change. Z/IPStream X/2 supports Microsoft’s Smooth Streaming and Apple HLS
adaptive streaming technologies, encoding the same stream at multiple bitrates and keeping audio
packets sample-aligned. Adaptive Streaming ensures that your listeners are automatically receiving
optimal quality and consistency based on the bandwidth of their connection.
Audio Replacement/Blanking
It is not uncommon for certain programming to be blacked out or contractually restricted from
streaming online. Z/IPStream X/2 makes quick work of programming blackouts by enabling you to
replace restricted material with content from a separate audio source, or audio from files. You have full
control over the switch points and the duration, and the switch points are sample-accurate when using
timestamped RTP audio for input.
Stream Synchronization
Stream synchronization is essential when implementing resilient streaming. Using Stream
Synchronization, separate encoder instances (running on different PCs and even at different locations)
are able to synchronize so that bitstreams generated by all instances are identical. This enables
resilient streaming deployment through redundancy. If one encoder goes down (or is taken down for
maintenance), the other encoder(s) continue to generate the appropriate stream, with no interruptions
to service. Timestamped RTP input and Smooth Streaming for output are required to use Stream
Synchronization.
Direct Livewire and RTP Audio Input
Z/IPStream X/2 works seamlessly with native Livewire audio sources, and can also accept RTP unicast
sources.
SNMP Alarms
Z/IPStream X/2 can be monitored via SNMP, a feature particularly important for large-scale
deployments. SNMP monitoring gives you peace of mind that your stream is fully functional, and if
anything does go wrong, SNMP alarms will detect and immediately inform you of any problems.
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Z/IPSTREAM X/2
REST API
In addition to an HTML5 web interface, Z/IPStream X/2 provides full programmatic control over its
functions. Customers can use the REST API for configuration, monitoring, or dynamic control. REST is a
web standard that can be used with the majority of scripting or programming languages from JavaScript
to Python, Ruby, and more. Z/IPStream X/2 gives you complete control of your stream through a variety
of industry-standard interfaces.
Cloud-Ready
Z/IPStream X/2 is a software-only application that’s cloud-ready. It is designed to run in the background
as a Windows service, and its HTML5 web interface makes remote configuration a breeze from PCs,
Macs, tablets, or even smart phones. The REST API is ready to handle any additional custom control or
monitoring requirements. Whether off-site or on, Z/IPStream X/2 gives you the flexibility to set up your
stream however it best suits your needs.
Z/IPStream 9X/2: Full Omnia.9 Audio Processing
Z/IPStream X/2 can be upgraded to Z/IPStream 9X/2 at any time. Z/IPStream 9X/2 takes the already
rock-solid 3-Band Omnia processing in the X/2 and elevates it with full Omnia.9 audio processing by Leif
Claesson, which includes exclusive “Undo” Technology, the full Omnia.9 toolbox, and much more.
SPECIFICATIONS
• Windows 7 or later OS, 32-bit or 64-bit version
• 1 gigahertz (GHz) or faster 32-bit (x86) or 64-bit (x64) processor
• 1 gigabyte (GB) RAM (32-bit) or 2 GB RAM (64-bit)
• 200 MB free disk space required for installation
• Additional disk space is used for logging
• Internet access
• Administrative privileges required during installation
• Web browser required for configuration and management
• When AES67 is used as input, only stereo mode is supported
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Z/IPSTREAM 9X/2
Z/IPStream 9X/2
The ultimate, high-quality processing/encoding
software with proprietary audio correction and
sonic management
OVERVIEW
Based on the technology found in the popular Omnia.9 audio processor, 9X/2 is not simply a streaming
processor-encoder, but a complete audio management system that will actually improve the flaws found
in most recorded source material – both music and voice – as well as address the specific technical
challenges of Internet distribution.
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Z/IPSTREAM 9X/2
FEATURES
• Exclusive “Undo” technology with de-clipper prevents listener fatigue by removing distortion and
selectively, undoing the over-compression so common in mastering today
• Optimizes sound quality of low bitrates by removing distortion components so that they do not waste
bits during encoding.
• 6-band Parametric EQ for your signature sound
• Downward Expansion (source noise reduction)
• Multiband stereo enhancer
Additional Features
• Software only, no special cards required
• Includes Virtual Audio Cable to receive audio from other programs on the same machine
• Includes Axia Livewire® driver
• Runs as a Windows service in the background, no need to log in
• Manage from anywhere with NfRemote, locally or across the Internet
• Up to 16 fully independent stereo processors in one instance, and up to 8 instances on one machine.
Pay only for what you need. Upgrades available
• Local monitor output with patch-point selection and full speaker controller
• Flexible remote control application with touch screen support, comprehensive instrumentation, and
remote audio streaming of any patch-point, also includes full speaker controller
• Separately adjustable sample rate (high-quality conversion) and gain control per encoded stream
• Extremely high audio quality, efficient CPU usage and low memory footprint
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Z/IPSTREAM 9X/2
IN DEPTH
Z/IPStream 9X/2 comes with both a GUI application and a service that contain the exact same
processing. During initial set-up (sound card configuration, etc.), use the GUI application. Once initial
configuration is done and tested, switch over to using the Service, which you can then control with
NfRemote from any computer.
Everything can be controlled with NfRemote except for which sound cards to use. Z/IPStream 9X/2 and
NfRemote are standard Windows 32-bit native applications and do not use Microsoft.NET or similar.
Z/IPStream 9X/2 is primarily designed for streaming and only has one local sound card output. However,
NfRemote has built-in dedicated PCM audio streaming for monitoring, so that you can monitor with low
delay from any computer, for example, while adjusting the processing.
9X/2 can encode audio to MP3, AAC, HE-AAC v1/v2 h (aacPlus), MP2, and WMA. Low complexity AAC
(AAC-LC), high-efficiency AAC (HE-AAC), and extended HE-AAC (xHE-AAC) are all supported. AAC has
been standardized under both MPEG-2 and MPEG-4. The format most commonly used is MPEG-4
AAC-LC. Often this is called just ‘AAC’. HE-AAC adds Spectral Band Replication to AAC and it is sometimes
called AAC+ (sometimes seen as ‘aacPlus’ or ‘AACplus’). There is also an HE-AAC v2 format which adds
parametric stereo optimizations to HE-AAC. Sometimes this is called AAC+ v2 or Enhanced AAC+. Finally,
xHE-AAC, the latest Fraunhofer codec, works well at low bitrates and therefore has more encoding
power. Whereas, other codecs like AAC and MP3 sound much better for music than they do for speech,
xHE-AAC sounds great for both speech and music, even at the lowest bitrates. 9X/2 can also use
Windows Media codecs installed on the system, 48kbps or higher.
9X/2 can directly feed SHOUTcast-style servers (SHOUTcast, Icecast, Steamcast, etc.). The Wowza
server is also supported for streaming to Flash clients. Windows Media streams can be sent to
Windows Media server.
A few words about Undo
Undo is two stages:
First, the de-clipper removes distortion by detecting clipped edges of the waveform and resynthesizing
the missing part. Unlike simpler algorithms, no distortion is ever created as the resynthesizing is
performed entirely in frequency domain.
Second, the amount of short-term dynamics is detected for each of 5 frequency bands, and
automatically controls the threshold and expansion ratio of 5 upwards expanders, to undo excessive
compression and peak limiting.
Both techniques together result in an incredible “is that really the same recording” level of improvement.
Audio quality of low bitrate codecs is also vastly improved, as a less distorted waveform is less
complicated for the codec to encode (thus using fewer bits) and more dynamic, punchy sound gives the
codec a place to hide the bitrate reduction artifacts.
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Z/IPSTREAM 9X/2
SPECIFICATIONS
System requirements
• 9X/2 will run on Windows XP or newer. Minimum requirements are Core 2 Duo, 512 MB RAM
General
• A Core i7 2600 and 4 GB RAM comfortably runs 16 stereo processors with several encoders each
• Supports multiple ASIO and WDM (Wave/Direct-Sound/Kernel Streaming) audio interfaces
simultaneously. Input selection can be done on the fly
• Simultaneous MP3/AAC/aacPlus/MP2/WMA encoding, compatible with Shoutcast, Icecast, Wowza,
and Windows Media server
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Z/IPSTREAM A/XE
Z/IPStream A/XE
Entry level processing and encoding software
for streaming
OVERVIEW
Z/IPStream A/XE can process audio for a variety of applications, bitrate-reduced and linear. It runs in the
background as a Windows service, can be fully-managed and configured remotely with a web browser,
and can even process and encode multiple streams in various formats simultaneously.
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Z/IPSTREAM A/XE
FEATURES
• Genuine 3-Band Omnia processing to improve audio levels, loudness and perceived quality.
• Software only, no special cards required
• Runs as a Windows service in the background. No need to log in.
• Managed from anywhere through a web browser, locally or across the Internet
• Each license = one stereo input. The user can add each license to the same PC or separate PCs.
• Each program input can be processed and encoded in multiple ways, and sent to multiple servers
simultaneously.
• Processed audio can also be sent to a local sound device for monitoring.
Additional Features
• High-performance, low memory footprint, native application
• Can operate with Virtual Audio Cable driver. This allows A/XE to accept audio from a playout system
or other applications on the same PC. It can also be used to feed the processed audio from A/XE to
another application on the same PC.
• All configuration information is stored in a single XML file for simple configuration backup/restore.
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Z/IPSTREAM A/XE
IN DEPTH
Seamlessly integrates with other software
The new Virtual Audio Cable allows Z/IPStream A/XE to receive, process, and send audio to other
software on the PC. Internally encoded Shoutcast or Wowza server streams can be “tagged” with “nowplaying” information received from automation systems or another application. An internal scheduler can
start and stop streams at specific times, or daypart processing presets for different shows.
Included with A/XE is a license for the multi-channel version of the Axia IP-Audio driver. Customers
with a Livewire® installation can use the Axia IP-Audio driver to send and receive audio directly from a
Livewire network without the need for hardware audio cards.
Metadata
Accepts metadata from a variety of sources and uses it to “tag” the audio stream. This information is
then sent to the media server and displayed in the user’s media player. Metadata can be accepted over
TCP/IP and UDP, or from text files. A/XE can accept just about any format, from simple, line-based
messages to XML messages and anything in between. An included set of metadata filters (small scripts,
using the Lua scripting language) can be further edited and customized.
Processing by Omnia
Z/IPStream A/XE features adjustable wide-band AGC with a three-band compressor/limiter, IIF EQ and
low-pass filter, and a precision look-ahead final limiter to prevent clipping. Resulting streams are cleaner,
clearer, and with more presence and detail.
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Z/IPSTREAM A/XE
SPECIFICATIONS
Hardware Requirements
• 32-bit Windows XP and later
• Minimum 512MB RAM
• 20MB free hard-drive space
• Network Interface Card
Codecs
• MP3, AAC, HE-AAC, HE-AAC v2. The highest quality codecs from Fraunhofer
Streaming Servers Supported
• ShoutCAST-compatible servers, including ShoutCAST v2
• Icecast
• Adobe Flash Media server
• Wowza server
• Live365
• Windows Media Server
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Z/IPSTREAM F/XE
Z/IPStream F/XE
Processing/encoding software for
file-based material
OVERVIEW
Combines Omnia® audio processing with the Fraunhofer MP3 and AAC codecs for high quality
processing preparation for podcasting or file based streaming/encoding.
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Z/IPSTREAM F/XE
FEATURES
• Genuine Omnia processing to improve audio levels, loudness and perceived quality.
• Software only, no special cards required
• Able to read PCM WAV files, MPEG Layer-2 and MPEG Layer-3 source files.
• Can automatically send the output file to an FTP server.
• Can notify the user by email if problems are detected
• Logs are kept during processing so you can find the source of a problem
Additional Features
• Read metadata from external files and embed the information as ID3 tags in the output files.
• Encode the output audio using MP3 or AAC (including HE AAC and HE AAC v2), or save linear PCM WAV
audio files.
• Core processing and encoding uses high-performance, low memory footprint, native application
• Drop files on FileProcessor for on-demand processing and encoding, or automate your work using
FolderBot to watch folders for new files and automatically process them as they arrive.
• You can define multiple configurations in FileProcessor. Each configuration can process and encode
the files with a different set of parameters or send the output to different locations. This makes it
easy to define and reuse project-specific configurations.
• FolderBot watches one or more folders and automatically processes the files as they are added to the
folder. Files can be handled differently based on the watched folder.
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Z/IPSTREAM F/XE
IN DEPTH
Metadata
Z/IPStream F/XE will read metadata from external files and embed the information as ID3 tags in
the output files. The core processing and encoding uses high-performance, low memory footprint,
native application. You can use drop files on FileProcessor for on-demand processing and encoding, or
automate your work using FolderBot to watch folders for new files and automatically process them
as they arrive. Multiple configurations are able to be defined in FileProcessor. Each configuration can
process and encode the files with a different set of parameters or send the output to different locations.
This makes it easy to define and reuse project-specific configurations.
Included with F/XE
F/XE includes a license to the multi-channel version of the Axia IP-Audio driver. Customers with a
Livewire® installation can use the Axia IP-Audio driver to send and receive audio directly from the
network without the need for hardware audio cards.
SPECIFICATIONS
System Requirements
• Windows XP or later with 20MB of free disk space
• Microsoft .NET client framework 4.0
• Internet access
Codecs
• MP3, AAC, HE-AAC, HE-AAC v2. The highest quality codecs from Fraunhofer
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Z/IPSTREAM R/1
Z/IPStream R/1
The professional, one-box streaming appliance
OVERVIEW
Z/IPStream combines audio processing with MP3 and AAC encoding in one convenient, single-rack
unit. The AAC encoder supports AAC-LC, HE-AAC and HE-AAC v2 formats, and is fully managed and
configured remotely with any standard Web browser. Z/IPStream features a wideband AGC, 3-band
compressor/limiter, EQ, low-pass filter and a precision look-ahead final limiter; processed audio can then
be encoded directly to MP3 or AAC streams to feed a remote replication server at your ISP. Streams can
be “tagged” with “now-playing” information received from automation systems. Analog and Livewire®
I/O are standard.
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Z/IPSTREAM R/1
FEATURES
• On-card audio processing includes wideband AGC, 4-band EQ and tone effects
• Includes a stereo enhancer, 3-band AGC with Fidelity control and SIS™ (Sound Impact System) to
manage spectral balance
• 4-band limiter and brick-wall final limiter.
• Bit-reduced encoding is handled by host PC CPU.
• Supports MP3, AAC, and HE-AAC v1/v2 encoding, plus 3GP-compatible encoding for mobile phones.
• Compatible with all standard streaming server platforms including Darwin, Flash, Helix, Icecast 2,
Red5, Shoutcast and Wowza using HTTP/ICY, RTSP/RTP Unicast, and RTMP protocols.
• Accepts audio input via Axia Livewire® connection or PCI (WDM driver).
• Four and eight channel models.
Additional Features
• Audio pre-processing, stream encoding and delivery to remote replication server, all in a professional
1RU appliance.
• Pro-grade 24-bit A/D converter for studio-reference quality audio.
• Choice of MP3 or AAC-LC, HE-AAC, HE-AAC v2 stream coding, with output bit rates from 16 kbps to
320 kbps (dependent upon active codec).
• Omnia audio processing includes wideband AGC, 3-band compressor/limiter, EQ, low-pass filter and
precision look-ahead final limiter.
• Metadata support for all popular playout platforms allows streams to be dynamically tagged with
“now-playing” information from automation systems.
• Studio-grade analog and Livewire IP-Audio I/O, with separate LAN & WAN Ethernet ports.
• Directly supports ICEcast, SHOUTcast, SHOUTcast v2, Adobe Flash Media Server as well as Adobe
RTMP, RTP streams (including RTP multicast), as well as LimeLight, Akamai and other popular
streaming servers.
• Dual encoder support can be used to provide high and low bitrate streams, or MP3 and AAC at the
same time.
• Can accept metadata over RS-232 (using USB to RS232 adapter).
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Z/IPSTREAM R/1
IN DEPTH
Plug. Play. Stream.
For years, the way to stream audio to Internet listeners included unbalanced mini-jacks, poor-quality
sound cards, one or more PCs to maintain, and a collection of software that didn’t always play nicely
together. Broadcasters asked for a professional, PC-free Web streaming solution — and Telos delivers.
Z/IPStream R/1 takes the hassle out of streaming. There’s no PC needed; Z/IPStream R/1 takes just
1RU of rack space. Slide it in and it’s ready to gostreaming. Just send audio to Z/IPStream R/1, make a
few setup selections and, within minutes, you’ll be streaming your programming to your favorite stream
server or streaming service for worldwide distribution.
Broadcasters know that Telos is the codec expert, and Omnia is the processing authority. Z/IPStream
R/1 puts all of our expertise into one integrated streaming appliance. First, incoming audio gets
treated to pre-processing from Omnia, using algorithms that work hand-in-glove with Z/IPStream
R/1’s codecs to shape and optimize audio prior to encoding. Then, genuine MPEG encoding algorithms
from FhG, the inventors of MP3, ensure the most artifact-free sound quality at whatever bit rate you
choose. Encode directly to an MP3 or MPEG-AAC stream, then send it to a Shoutcast, Wowza, Icecast,
LimeLight, Akamai, Adobe Flash Media server, or other popular streaming server for distribution to
your waiting listeners.
Setup is a breeze. Log in with a laptop and Web browser for easy setup or remote control, or tweak
the front-panel controls. There’s also a convenient built-in headphone amp with 1/4” jack and volume
control for last minute in-the-rack fine tuning.
Z/IPStream R/1 comes with studio-grade analog inputs and outputs, plus Livewire Audio over IP. On
the output side, Z/IPStream R/1 delivers fully processed, unencoded audio as well as encoded audio,
providing your studio with another source for processed sound. Full network connectivity is provided via
two Ethernet jacks, one for the LAN (including Livewire) and the other for the WAN and streaming.
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Z/IPSTREAM R/1
The Professional Choice for Streaming Audio.
Optimizing sound quality is as essential on the web as it is on traditional formats. Z/IPStream R/1 has
a built-in processing section that works together with the streaming encoder, optimizing your audio
for stunning sound — even after bit-reduction. This isn’t just some cheap leveler – it’s real processing
by Omnia, complete with wideband AGC, a 3-band combined compressor/limiter, high-frequency EQ,
an adjustable-bandwidth low-pass filter, and Omnia’s famous anti-aliasing final Look-Ahead limiter.
There are even a selection of presets, tailored to specific formats and bit rates, to help you get up
and running quickly.
Of course, the foundation for high fidelity audio distribution rests on professional encoding technology.
The quality of the encoder directly affects the quality of the output. Telos has a long history of
partnership with Germany’s Fraunhofer Gesellschaft Laboratory (FhG), the world leader in audio
compression research and the inventors of MP3; Z/IPStream R/1 uses genuine MP3 and MPEG-AAC
encoding algorithms to ensure the most artifact-free sound quality at any bit rate you choose, from 16
kbps all the way to 320 kbps. No other encoder has this pedigree, or achieves this level of quality and
performance. Generic “mp3” encoders can’t come close.
Z/IPStream R/1 gives you a wide choice of genuine Fraunhofer encoding algorithms, which include
MP3, the Standard for digital audio. It’s the safest codec choice for compatibility with the widest variety
of listening devices. Or choose AAC-LC, a high performance codec for excellent audio quality at lower
bitrates. AAC-LC is in widespread use, most notably in Apple’s iTunes. And then there’s High Efficiency
Advanced Audio Coding, or HE-AAC, a newer AAC codec which incorporates Spectral Band Replication
(SBR) bandwidth expansion to improve audio at very low bitrates. HE-AAC v2 applies a Parametric
Stereo feature to HE-AAC codec allowing for even further reduction in bandwidth.
When you’re done processing and encoding, select your metadata source and feed your stream to any
SHOUTcast or SHOUTcast v2-compatible media server, or a Wowza server for streaming to Flash clients.
Z/IPStream R/1 works with ICECast and Adobe Flash Media and Adobe RTMP servers too, as well as
popular streaming services from LimeLight, Akamai, and other popular streaming service providers. You
can feed directly to a streaming server on your LAN, to an Internet streaming relay service via the WAN
port, or take processed audio from the rear-panel XLR outputs. No matter what your audio source or
how you stream, Z/IPStream R/1 delivers flawlessly optimized audio that sounds terrific.
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Z/IPSTREAM R/1
SPECIFICATIONS
Audio Coding
Codec Choices:
• MP3: 16 to 320 kbps
• AAC-LC: 24 to 320 kbps
• HE-AAC: 24-96 kbps
• HE-AAC v2 (aacPlus): 24-96 kbps
AAC Transport Modes:
• ADTS
• ADTS-CRC
• ADIF
• RAW
Metadata Formats:
• Character Parser Sample
• Line Parser Sample
• Nexgen Audio Sense
• Simian Template 1
• XML Parser Sample
• XML-Jazler
• XML-Jazler2
• XML-MediaTouch
• XML-MediaTouch2
• XML-Sample2
• XML-Zetta
• User-definable
Input
• Analog: Balanced XLR, +4 dBu
• Input Impedance: 6K Ohm differential
• Analog to Digital Converter: 24bits
• Digital: Livewire AoIP, via LAN port
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Z/IPSTREAM R/1
Output
• Analog: Balanced XLR
• Output Clipping: + 22dBu
• Output Impedance: 50 Ohm differential
• Digital to Analog Converter: 24bits
• Digital: Livewire AoIP, via LAN or WAN port
Audio Performance
• THD+N: < 0.03% @ +12dBu, 1 kHz Sine
• Frequency Response: +/- 1dB 25– 20 kHz
• Headroom: 18dB
• Dynamic Range: > 87dB Unweighted > 90 dB “A” Weighted
• Crosstalk: > 80 db
Remote Control
• LAN via built-in Webserver
Power
• Internal supply, 85–250 VAC auto-switching, 50–60 Hz
• Power consumption: 14.2 Watts
Regulatory
North America: FCC and CE tested and compliant, power supply is UL approved.
Europe: Complies with the European Union Directive 2002/95/EC on the restriction of the use of certain
hazardous substances in electrical and electronic equipment (RoHS), as amended by Commission
Decisions 2005/618/EC, 2005/717/ EC, 2005/747/EC (RoHS Directive), and WEEE.
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Z/IPSTREAM R/2
Z/IPStream R/2™
Stream Encoder/Processor
The best-sounding streams...from the smallest box.
OVERVIEW
Processing and Encoding of Multiple Programs
Z/IPStream R/2 produces the best possible streams by providing a multitude of streaming options for the
broadcaster and maximizing audio quality for the listener. This second-generation Z/IPStream processor
and encoder is essentially the hardware appliance version of the successful X/2 and 9X/2 software,
allowing flexible, multi-format stream-encoding for up to eight audio programs in a single 1RU chassis.
Ideal for high-density processing and encoding applications, R/2 offers the simplicity and reliability of a
single 1RU dedicated hardware appliance. R/2 is available with 3-band Omnia processing or full Omnia.9
processing, both featuring high-quality Telos® encoding.
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Z/IPSTREAM R/2
FEATURES
• Processing and stream encoding of up to eight audio programs in 1RU
• Available in models with 3-band or full Omnia.9 processing
• AES/EBU and Livewire®/AES67 audio I/O
• Encode a program at multiple bitrates for adaptive streaming applications. Apple HLS and Microsoft
Smooth Streaming formats are supported
• AAC-LC, HE-AAC, HE-AAC v2, xHE-AAC and MP3 stream encoding from 16 kbps to 320 kbps
depending on codec used
• xHE-AAC for low-bitrate streaming
• Dual power supplies and dual gigabit Ethernet ports for reliable, 24/7 operation
• Processing or encoding can be used independently if desired
• Process and encode the same audio program in multiple formats. Simultaneously send the encoded
streams to multiple destinations
• Supported server platforms include ICEcast, SHOUTcast, SHOUTcast v2, Adobe Flash Media Server,
Wowza, as well as Triton Digital, LimeLight, Akamai, and other popular streaming services
• Includes support for RTP and RTP multicast streams
• Built-in HTTP server can directly serve HLS streams
• HTML5 web-based remote control for administration
• SNMP support allows direct monitoring from your SNMP management system, or you can receive
alerts via email
• Integrates into your workflow: REST-ful API allows full control from your application to start/
stop streams, switch audio sources or insert audio content from files; or monitor multiple devices
simultaneously
• Dedicated IP remote control software with test instrumentation (RTA, FFT, oscilloscopes, loudness
metering) for audio-processing adjustments when using Omnia.9 processing
• New flexible Metadata allows R/2 to accept metadata from multiple play-out systems and lets
broadcasters tweak the fields they want to present to listeners
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Z/IPSTREAM R/2
IN DEPTH
Z/IPStream R/2 is the latest generation of streaming audio processing and encoding hardware in the
Z/IPStream family, handling processing and encoding of multiple audio programs in a compact 1 RU
chassis. Processing and encoding of up to eight audio programs is supported, with Livewire and AES/
EBU I/O audio input.
The base unit includes processing and encoding of two audio programs using the standard 3-band
Omnia audio processing. Encoding formats including MP3, AAC-LC, HE-AAC, HE-AAC v2, and xHE-AAC.
Multiple codecs and bitrates are supported simultaneously on each audio program. A special multirate
AAC encoder is included for adaptive bitrate streaming applications. Supported streaming platforms include
ICEcast, SHOUTcast, SHOUTcast v2, Adobe Flash Media Server, Adobe RTMP, Triton Digital, LimeLight,
Akamai, and Wowza. Additional audio program inputs and Omnia.9 processing are available as options.
The Z/IPStream R/2 with Omnia.9 processing models include exclusive ‘Undo’ de-clipping, 6-band
parametric EQ, downward expansion (source noise reduction), multiband stereo enhancer, up to threestage AGC with adjustable sidechain filter, 2-7 bands of processing, and final two-band look-ahead
limiter. Full IP remote control of processing parameters is available via NFRemote, along with the
complete Omnia.9 suite of test instrumentation (loudness metering, FFT, RTA, oscilloscope, and remote
client audio streaming).
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Z/IPSTREAM R/2
SPECIFICATIONS
Processing
• Includes standard 3-band Omnia processing. Optionally use full Omnia.9 processing with up to 7
bands of processing. The number of audio processing instances that may be used simultaneously
depends on overall system configuration and resource usage. As expected, Omnia.9 is more resourceintensive than the 3-band Omnia processor. The chart below illustrates the number of instances that
can be run under typical usage scenarios. It is provided as a guide, the actual number may be different
for your specific application.
Stream Encoding
• Includes AAC-LC, HE-AAC, HE-AAC v2, xHE-AAC, and MP3 encoding at bitrates from 16 kbps up to
320 kbps (depending on codec). A program may be encoded using multiple codec formats and bitrates
simultaneously. A special multirate encoder supports encoding for adaptive streaming applications.
The multirate encoder properly generates the required Stream Access Points for adaptive streaming.
Ethernet Remote Control
• Gigabit Ethernet supports HTML web interface for administration, REST API for remote control, and
SNMP monitoring. Also used with dedicated remote control application for Omnia.9 processing.
Various metadata update methods via Ethernet supported as well.
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Z/IPSTREAM R/2
Front Panel Controls and Indicators
• Directional navigation cluster
• Graphical LCD display
• Power/Reset controls
• Diagnostic LED indicators (power, network, drive activity)
• USB port
Audio I/O
• Livewire/AES67 and AES/EBU audio I/O
• Supports AES/EBU input at up to 24 bits, 192 kHz
• Supports direct input from RTP streams
Power Requirements
Dual power supplies, each rated at 100-264 VAC, 50/60Hz, auto-sensing, 100W max total
Dimensions and Weight
• One rack unit— 1.75”H x 19”W x 15.5”D (44 x 483 x 394 mm)
• Net weight: 9 lbs (4 kg); shipping: 12 lbs (5.4 kg) approximate
Environmental
• Fan cooled
• Operating: 0 to 50 degrees C
• Non-operating: –20 to 70 degrees C
Regulatory
North America: FCC and CE tested and compliant. Power supply is UL approved.
Europe: Complies with the European Union Directive 2002/95/EC on the restriction of the use of certain
hazardous substances in electrical and electronic equipment (RoHS), as amended by Commission
Decisions 2005/618/EC, 2005/717/ EC, 2005/747/EC (RoHS Directive), and WEEE.
Warranty
Standard Telos Alliance 5-Year Warranty
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LINEAR ACOUSTIC | AERO.2000
AERO.2000™
Audio/Loudness Manager
OVERVIEW
Like AERO.100™, AERO.2000 includes a single instance of AEROMAX® processing in your choice
of AMX5.1 (5.1+2+2), AMX2.0 (2+2+2), or AMX 5x2 (2+2+2+2+2) configurations plus upmixing/
downmixing via our UPMAX®II algorithm – but adds the flexibility and convenience of a full color display,
menu navigation controls, and a headphone output to the front panel.
A second AMX5.1, AMX2.0, or AMX5.2 instance is optionally available. Dolby® decoding, encoding, and
Nielsen® watermark encoding are offered as options either or both instances.
I/O includes de-embedding and re-embedding of eight pairs of HD/SD-SDI audio just like AERO.100, but
AERO.2000 supports a full eight pairs of AES audio.
Support for SAP/DVS, EAS, local emergency audio, local voiceover, and optionally, Audio Description
(warble tone) functionality is included along with a new scheduling feature and task scripting for
enhanced GPI functionality. CrowdControl™ is now standard for increased dialogue intelligibility.
ITU-R BS.1770-3 metering and logging (including True Peak) is included for all program outputs with
displays for LKFS or EBU R-128. NfRemote software is included for remote configuration, control and
metering over an Ethernet connection while a built-in HTTP server enables control of I/O, presets,
and individual processing parameters using simple IP commands. Compensating video delay and dual
redundant internal power supplies are standard.
DTV AUDIO PERFECTION FROM PRODUCTION TO TRANSMISSION
TELOSALLIANCE.COM
LINEAR ACOUSTIC | AERO.2000
FEATURES
• Linear Acoustic AEROMAX® loudness and dynamic range control
• UPMAX® II upmixing with AutoMAX-II™
• Linear Acoustic Intelligent Dynamics® with advanced ITU (AI) Limiter
• Standard single processing instance in AMX5.1 (5.1+2+2), AMX2.0 (2+2+2), or
AMX 5x2 (2+2+2+2+2) configuration
• Available second instance in AMX5.1, AMX2.0, or AMX5.2 configuration
• Automatic downmix output
• Standard CrowdControl TM for increased dialog intelligibility
• Support for SAP/DVS
• Local audio/voiceover insertion
• Optional Audio Description (warble tone)
• Available Dolby Digital (AC-3), Dolby Digital Plus and Dolby E decoding
• Available Dolby Digital (AC-3), Dolby Digital Plus, and Dolby E encoding
• HD/SD-SDI I/O with included video delay (16 channels)
• AES I/O with reference input (16 channels)
• Dual PSU and relay bypass
• Extensive TCP/IP remote control and HTTP control
DTV AUDIO PERFECTION FROM PRODUCTION TO TRANSMISSION
TELOSALLIANCE.COM
LINEAR ACOUSTIC | AERO.2000
IN DEPTH
Comprehensive TCP/IP Remote
Provides control over all system settings, processing and coding parameters plus extensive metering
of signal presence, processing and coding activity, and audio loudness. System status reports physical
I/O details along with system, power supply and environmental health. The remote application also delivers
remote audio, including 5.1 channels, to the user so that signal quality can be auditioned anywhere link
bandwidth permits. The built-in HTTP server provides for simple get/set control of all parameters and status.
Nielsen® Encoding
Generates revenue-critical Nielsen NAES II and the new Nielsen Watermarks audience measurement
codes. AERO.2000 precisely inserts these signals for maximum code recovery – after audio decoding
and processing and before transmission encoding.
Linear Acoustic Intelligent Dynamics®
The Basics
Linear Acoustic Intelligent Dynamics hybrid metadata processing is a patented hybrid of traditional
multiband techniques and metadata. Because it is created during transmission encoding, this metadata
requires no operator intervention or special tools and is a new version of the DRC part of the Dolby
Digital encoder that has always been there. Except it is now effective and uncomplicated.
Featured at no additional cost, Intelligent Dynamics is actually a combination of two revolutionary
technologies.
DTV AUDIO PERFECTION FROM PRODUCTION TO TRANSMISSION
TELOSALLIANCE.COM
LINEAR ACOUSTIC | AERO.2000
How it Works
The first portion of Intelligent Dynamics is the Emmy® award-winning process developed by Linear
Acoustic which generates new metadata dynamic range control via traditional processing techniques.
This control can be set as permanent, reversible, or anywhere in between. This technology is already
present in every AERO.100, AERO.2000, AERO.asi and AERO.soft product.
The second portion of the process involves verifying and marking content as compliant upstream. This
is accomplished using a portion of the new Dolby Intelligent Media Framework called Evolution which
enables measurement and incorporation of authenticated loudness data within the audio at each
stage of the content delivery chain from production onwards. Compliant programming can be passed
through with minimal or no additional processing, while content that cannot be verified can still be made
compliant.
Most current television audio processors control loudness by managing dynamic range in real time. This
effectively manages “boundary issues” such as commercials that follow a quiet program segment, but
also compromises the impact and excitement of intentionally dynamic scenes because all programming
gets some degree of permanent correction.
Audio transmission formats such as Dolby Digital (AC-3) and Dolby Digital Plus (E-AC-3) already
incorporate metadata dynamic range control (DRC) profiles, which can provide some control, but
these profiles are not ideally suited for broadcast audio and often result in significant over and under
processing if other metadata parameters are not correct.
Intelligent Dynamics overcomes these compromises and enables audio to be processed to the degree
and permanence dictated by the programming itself. The content effectively and automatically controls
the processing and eliminates the need for further downstream changes that could irreparably change it.
The Result
Consumers can now enjoy the benefits of audio that is tailored to their liking using existing Dolby Digital
and Dolby Digital Plus decoders. From the default of controlled dynamic range for noisy environments
and small television speakers to full dynamic range for well-produced programming and multi-channel
home theater systems, Linear Acoustic Intelligent Dynamics delivers the best of both worlds without
compromise.
DTV AUDIO PERFECTION FROM PRODUCTION TO TRANSMISSION
TELOSALLIANCE.COM
LINEAR ACOUSTIC | AERO.2000
SPECIFICATIONS
Processing
• AEROMAX multistage adaptive wideband and multiband loudness and dynamic range control with
ITU-R BS.1770 and EBU R128 loudness metering.
• Dual UPMAX II two-channel to 5.1-channel upmixers plus main channel downmixing with automatic
bypass of discrete content.
• Linear Acoustic Intelligent Dynamics with Advanced ITU (AI) Limiter
Audio Encoding/Decoding
• Available Dolby Digital (AC-3), Dolby Digital Plus and Dolby E decoding
• Available Dolby Digital (AC-3), Dolby Digital Plus, and Dolby E encoding
Reference
• 48kHz via AES DARS (or any AES signal applied to the Ref In connector), AES In 1, SDI, or from the
internal 48kHz clock (standalone use only).
Sample Rate/Resolution/Frequency Response
• 48kHz, 24-bit, 20Hz to 20kHz below threshold
AES I/O
• Eight main inputs plus reference via 75-Ohm BNC female connectors. Eight main outputs plus encoder
output. Eight additional channels of auxiliary digital I/O on DB-25 female connector. All digital inputs
are 75 Ohm internally terminated, unbalanced. Signal levels per SMPTE 276M/ AES-3ID-2001.
HD/SD-SDI I/O
• Auto sensing HD/SD-SDI intput up to 1080p/60/59.94/50Hz supported. De-embed up to 16 channels
from applied SDI signal, process and/or encode, re-embed up to 16 channels. Supports SMPTE 2020A
and B VANC metadata.
Headphone Output
• 1/4’’ (6.35mm) front panel connector with volume control.
DTV AUDIO PERFECTION FROM PRODUCTION TO TRANSMISSION
TELOSALLIANCE.COM
LINEAR ACOUSTIC | AERO.2000
GPI/O
• Parallel GPI/O Parallel Control Port
• 25-pin female D connector, 0-5V TTL levels for 8 inputs and 8 outputs; controls simple preset recalls
plus voiceover/EAS insertion
Serial Metadata Input
• 9-pin female D connector; 115.2 kbps; pinout per SMPTE 207M (RS-485); Designed to directly
interface with Dolby serial metadata (SMPTE RDD6)
Ethernet
• Gigabit Ethernet via RJ45 supports included TCP/IP remote control application; HTTP server included
for get/set control of all parameters.
Remote Control
• Windows®-compatible TCP/IP remote control Included application for full setup and control, ITU-R
BS.1770 metering for all programs, encoder statistics, and return audio for remote monitoring
(network speed permitting). HTTP server allows get/set control from PC Front Panel Controls and
Indicators.
Front Panel Controls
• Rotary encoder and control keys plus color display and headphone output.
Power Requirements
• Dual redundant power supplies, each rated at 100-264 VAC, auto-sensing, 50/60 Hz,
175W each maximum
Dimensions and Weight
• 2RU: 3.50”H x 19”W x 17”D (89mm X 483mm X 432mm)
• Net weight: 13 lbs. (5.9 kg), approximate.
Shipping Dimensions and Weight
• 22”W x 20”D x 9”H (559 x 508 x229 mm)
• Net weight: 18 lbs. (8.2 kg), approximate.
DTV AUDIO PERFECTION FROM PRODUCTION TO TRANSMISSION
TELOSALLIANCE.COM
LINEAR ACOUSTIC | AERO.2000
Environmental
• Fan cooled. Operating: 0 to 50 degrees C, non-operating -20 to 70 degrees C
Regulatory
North America: FCC and CE tested and compliant, power supply is UL approved.
Europe: Complies with the European Union Directive 2002/95/EC on the restriction of the use of certain
hazardous substances in electrical and electronic equipment (RoHS), as amended by Commission
Decisions 2005/618/EC, 2005/717/ EC, 2005/747/EC (RoHS Directive), and WEEE.
Warranty
Standard 2-year limited parts and labor
DTV AUDIO PERFECTION FROM PRODUCTION TO TRANSMISSION
TELOSALLIANCE.COM
LINEAR ACOUSTIC | AERO.100
AERO.100™
DTV Audio Processor
OVERVIEW
AERO.100 includes a single instance of AEROMAX® processing in your choice of AMX5.1 (5.1+2+2),
AMX2.0 (2+2+2), or AMX 5x2 (2+2+2+2+2) configurations plus upmixing/downmixing via our UPMAX®II
algorithm.
A second AMX5.1, AMX2.0, or AMX5.2 instance is optionally available. Dolby® decoding, encoding, and
Nielsen® watermark encoding are offered as options either or both instances.
I/O includes de-embedding and re-embedding of eight pairs of HD/SD-SDI audio and four pairs of AES audio.
Support for SAP/DVS, EAS, local emergency audio, local voiceover, and optionally, Audio Description
(warble tone) functionality is included along with a new scheduling feature and task scripting for
enhanced GPI functionality. CrowdControl™ is now standard for increased dialogue intelligibility.
ITU-R BS.1770-3 metering and logging (including True Peak) is included for all program outputs with
displays for LKFS or EBU R-128. NfRemote software is included for remote configuration, control and
metering over an Ethernet connection while a built-in HTTP server enables control of I/O, presets,
and individual processing parameters using simple IP commands. Compensating video delay and dual
redundant internal power supplies are standard.
DTV AUDIO PERFECTION FROM PRODUCTION TO TRANSMISSION
TELOSALLIANCE.COM
LINEAR ACOUSTIC | AERO.100
FEATURES
• Linear Acoustic AEROMAX® loudness and dynamics control
• UPMAX® II upmixing with AutoMAX-II™
• Linear Acoustic Intelligent Dynamics® with advanced ITU (AI) Limiter
• Standard single processing instance in AMX5.1 (5.1+2+2), AMX2.0 (2+2+2), or AMX 5x2 (2+2+2+2+2)
configuration
• Available second instance in AMX5.1, AMX2.0, or AMX5.2 configuration
• Automatic downmix output
• Standard CrowdControl TM for increased dialog intelligibility
• Support for SAP/DVS
• Local audio/voiceover insertion
• Optional Audio Description (warble tone)
• Available Dolby Digital (AC-3), Dolby Digital Plus and Dolby E decoding
• Available Dolby Digital (AC-3), Dolby Digital Plus, and Dolby E encoding
• HD/SD-SDI I/O with included video delay (16 channels)
• AES I/O with reference input (8 channels)
• Dual PSU and relay bypass
• Extensive TCP/IP remote control and HTTP control
DTV AUDIO PERFECTION FROM PRODUCTION TO TRANSMISSION
TELOSALLIANCE.COM
LINEAR ACOUSTIC | AERO.100
IN DEPTH
Comprehensive TCP/IP Remote
Provides control over all system settings, processing and coding parameters plus extensive metering of
signal presence, processing and coding activity, and audio loudness. System status reports physical I/O
details along with system, power supply and environmental health. The remote application also delivers
remote audio, up to 5.1 channels, to the user so that signal quality can be auditioned anywhere link
bandwidth permits. HTTP server is also included for simple get/set control of all parameters and status.
Nielsen Watermark Encoding
Generates revenue-critical Nielsen NAES II and the new Nielsen Watermark audience measurement
codes. AERO.100 precisely inserts these signals for maximum code recovery after audio decoding and
processing and before transmission encoding.
Linear Acoustic Intelligent Dynamics®
The Basics
Linear Acoustic Intelligent Dynamics hybrid metadata processing is a patented hybrid of traditional
multiband techniques and metadata. Because it is created during transmission encoding, this metadata
requires no operator intervention or special tools and is a new version of the DRC part of the Dolby
Digital encoder that has always been there. Except it is now effective and uncomplicated.
Featured at no additional cost, Intelligent Dynamics is actually a combination of two revolutionary
technologies.
DTV AUDIO PERFECTION FROM PRODUCTION TO TRANSMISSION
TELOSALLIANCE.COM
LINEAR ACOUSTIC | AERO.100
How it Works
The first portion of Intelligent Dynamics is the Emmy® award-winning process developed by Linear
Acoustic which generates new metadata dynamic range control via traditional processing techniques.
This control can be set as permanent, reversible, or anywhere in between. This technology is already
present in every AERO.100, AERO.2000, AERO.asi and AERO.soft product.
The second portion of the process involves verifying and marking content as compliant upstream. This
is accomplished using a portion of the new Dolby Intelligent Media Framework called Evolution which
enables measurement and incorporation of authenticated loudness data within the audio at each
stage of the content delivery chain from production onwards. Compliant programming can be passed
through with minimal or no additional processing, while content that cannot be verified can still be made
compliant.
Most current television audio processors control loudness by managing dynamic range in real time. This
effectively manages “boundary issues” such as commercials that follow a quiet program segment, but
also compromises the impact and excitement of intentionally dynamic scenes because all programming
gets some degree of permanent correction.
Audio transmission formats such as Dolby Digital (AC-3) and Dolby Digital Plus (E-AC-3) already
incorporate metadata dynamic range control (DRC) profiles, which can provide some control, but
these profiles are not ideally suited for broadcast audio and often result in significant over and under
processing if other metadata parameters are not correct.
Intelligent Dynamics overcomes these compromises and enables audio to be processed to the degree
and permanence dictated by the programming itself. The content effectively and automatically controls
the processing and eliminates the need for further downstream changes that could irreparably change it.
The Result
Consumers can now enjoy the benefits of audio that is tailored to their liking using existing Dolby Digital
and Dolby Digital Plus decoders. From the default of controlled dynamic range for noisy environments
and small television speakers to full dynamic range for well-produced programming and multi-channel
home theater systems, Linear Acoustic Intelligent Dynamics delivers the best of both worlds without
compromise.
DTV AUDIO PERFECTION FROM PRODUCTION TO TRANSMISSION
TELOSALLIANCE.COM
LINEAR ACOUSTIC | AERO.100
SPECIFICATIONS
Processing
• AEROMAX multistage adaptive wideband and multiband loudness and dynamic range control with
ITU-R BS.1770 and EBU R128 loudness metering
• Dual UPMAX II two-channel to 5.1-channel upmixers plus main channel downmixing with automatic
bypass of discrete content
• Linear Acoustic Intelligent Dynamics with Advanced ITU (AI) Limiter
Audio Encoding/Decoding
• Available Dolby Digital (AC-3), Dolby Digital Plus and Dolby E decoding
• Available Dolby Digital (AC-3), Dolby Digital Plus, and Dolby E encoding
Reference
• 48kHz via AES DARS (or any AES signal applied to the Ref In connector), AES In 1, SDI, or from the
internal 48kHz clock (standalone use only)
Sample Rate/Resolution/Frequency Response
• 48kHz, 24-bit, 20Hz to 20kHz below threshold
AES I/O
• Eight main inputs plus reference via 75-Ohm BNC female connectors. Eight main outputs plus encoder
output. All digital inputs are 75 Ohm internally terminated, unbalanced. Signal levels per SMPTE
276M/ AES-3ID-2001
HD/SD-SDI I/O
• Auto-sensing HD/SD-SDI (SMPTE 292M/259M) inputs, up to 1080i/60/59.94/50Hz. De-embed up
to 16 channels from applied SDI signal, process and/or encode, re-embed up to 16 channels. Supports
SMPTE 2020 A and B VANC metadata
Parallel GPI/O Control Port
• 25-pin female D connector, 0-5V TTL levels for 8 inputs and 8 outputs; controls simple preset recalls
plus voiceover/EAS insertion
Serial Metadata Input
• 9-pin female D connector; 115.2 kbps; pinout per SMPTE 207M (RS-485); Designed to directly
interface with Dolby serial metadata (SMPTE RDD6)
DTV AUDIO PERFECTION FROM PRODUCTION TO TRANSMISSION
TELOSALLIANCE.COM
LINEAR ACOUSTIC | AERO.100
Ethernet
• Gigabit Ethernet via RJ45 supports included TCP/IP remote control application; HTTP server included
for get/set control of all parameters
Remote Control
• Windows®-compatible TCP/IP remote control Included application for full setup and control, ITU-R
BS.1770 metering for all programs, encoder statistics, and return audio for remote monitoring
(network speed permitting). HTTP server allows get/set control from PC
Front Panel Controls and Indicators
• Graphical OLED display
Power Requirements
• Dual power supplies, each rated at 100-264 VAC, 50/60Hz, auto-sensing, 150W max. total
Dimensions and Weight
• 1RU - 1.75”H x 19”W x 15.5”D (44 x 483 x 394 mm)
• Net weight: 9 lbs. (4 kg), approximate.
Shipping Dimensions and Weight
• 22”W x 20”D x 7”H (559 x 508 x 178 mm)
• Net weight: 15 lbs. (6.80 kg), approximate.
Environmental
• Fan cooled. Operating: 0 to 50 degrees C, non-operating -20 to 70 degrees C
Regulatory
North America: FCC and CE tested and compliant, power supply is UL approved.
Europe: Complies with the European Union Directive 2002/95/EC on the restriction of the use of certain
hazardous substances in electrical and electronic equipment (RoHS), as amended by Commission
Decisions 2005/618/EC, 2005/717/ EC, 2005/747/EC (RoHS Directive), and WEEE.
Warranty
Standard 2-year limited parts and labor
DTV AUDIO PERFECTION FROM PRODUCTION TO TRANSMISSION
TELOSALLIANCE.COM
LINEAR ACOUSTIC | AERO.10
AERO.10
DTV Audio Processor
OVERVIEW
Highest Quality Television Audio Processing - Incredible Value
AERO.10 includes a single instance of AEROMAX® processing in your choice of AMX5.1 (5.1+2+2),
AMX2.0 (2+2+2), or AMX 5x2 (2+2+2+2+2) configurations plus upmixing/downmixing via our
UPMAX®II algorithm.
I/O includes de-embedding and re-embedding of eight pairs of HD/SD-SDI audio, four pairs of AES
audio, and balanced analogue stereo.
Support for SAP/DVS, EAS, local emergency audio, local voiceover, and optionally, Audio Description
(warble tone) functionality is included along with a new scheduling feature and task scripting for
enhanced GPI functionality.
ITU-R BS.1770-3 metering and logging (including True Peak) is included for all program outputs with
displays for LKFS or EBU R-128. NfRemote software is included for remote configuration, control and
metering over an Ethernet connection while a built-in HTTP server enables control of I/O, presets,
and individual processing parameters using simple IP commands. Compensating video delay and dual
redundant internal power supplies are standard.
DTV AUDIO PERFECTION FROM PRODUCTION TO TRANSMISSION
TELOSALLIANCE.COM
LINEAR ACOUSTIC | AERO.10
FEATURES
• Linear Acoustic AEROMAX® loudness and dynamics control
• UPMAX® II automatic upmixing and downmixing
• Advanced ITU (AI) Limiter
• Standard single processing instance in AMX5.1 (5.1+2+2), AMX2.0 (2+2+2), or
AMX 5x2 (2+2+2+2+2) configuration
• HD/SD-SDI I/O with included video delay
• 8 channels of AES I/O with reference input
• Balanced +4dBu stereo analog inputs and outputs
• Dual power supplies, autoranging for simple worldwide operation
• Relay bypass of all I/O
• Front panel GUI plus extensive TCP/IP and HTTP control
• Logging of loudness and True Peak data
DTV AUDIO PERFECTION FROM PRODUCTION TO TRANSMISSION
TELOSALLIANCE.COM
LINEAR ACOUSTIC | AERO.10
IN DEPTH
Highest quality industry standard audio control has never been more affordable. AERO.10 is a fullyfeatured audio processor supporting up to ten channels of audio and featuring a processing engine
identical to those in the AERO.100/1000/2000 products. Tools such as loudness and dynamic range
control, upmixing, downmixing, plus ITU and EBU compliant loudness metering and logging makes the
AERO.10 an extremely powerful solution for nearly any application at an extremely low cost.
To this, the AERO.10 adds a simple LCD front panel GUI and stereo analog I/O. The headphone output
has been designed to provide plenty of level even for difficult loads or quiet sources and is useful for
checking audio or adjusting processing.
New with AERO.10 is the addition of stereo analog inputs and outputs which serve to support facilities
amidst transition from analog to digital as well as interface with any analog device or signal path.
Comprehensive TCP/IP remote provides control over all system settings and processing parameters plus
extensive metering of loudness. System status reports physical I/O details along with system, power
supply and environmental health. The remote application also delivers remote audio, up to 5.1 channels,
so the user can audition signal quality anywhere link bandwidth permits. An HTTP server is also included
for simple get/set control of all parameters and retrieval of status and logging information. Constantly
active logging captures 7.5 day rolling weekly reports as well as specific time slots controlled by start/
stop. Loudness with multiple integration times as well as True Peak measurements are captured and
available for download.
Designed and built in the USA, the lightweight and rugged single rack-unit AERO.10 is a solid investment
in performance and flexibility. Though all current features are standard, future options can be enabled by
simply entering a factory provided key. Failover bypass relays on all I/O maintain signal continuity and
dual auto-ranging power supplies enable redundancy and worldwide compatibility.
DTV AUDIO PERFECTION FROM PRODUCTION TO TRANSMISSION
TELOSALLIANCE.COM
LINEAR ACOUSTIC | AERO.10
SPECIFICATIONS
Processing
• User selectable AEROMAX® processing in your choice of AMX5.1 (5.1+2+2), AMX2.0 (2+2+2), or AMX
5x2 (2+2+2+2+2) configurations
• Dual UPMAX II two-channel to 5.1 channel upmixers plus main channel downmixing, automatic
bypass of discrete content
• Linear Acoustic Advanced ITU (AI) Limiter
Sample Rate/Resolution/Frequency Response
• 48kHz, 24-bit, 20Hz to 20kHz below threshold
AES I/O
• Eight main inputs plus reference via 75-Ohm BNC female connectors, internally terminated; Eight
main outputs; Signal levels per SMPTE 276M/AES-3ID-2001
SDI I/O
• Auto-sensing HD/SD-SDI (SMPTE 292M/259M) inputs, up to 1080i/60/59.94/50Hz, access to audio
and VANC metadata
Analog I/O (stereo)
• 9-pin female D connector; 10K Ohm balanced stereo inputs; Balanced stereo outputs, +4dBu nominal,
+24dBu maximum into 600 Ohms.
Parallel GPI/O Control Port
• 25-pin female D connector, 0-5V TTL levels for 8 inputs and 8 outputs; controls simple preset recalls
plus voiceover/EAS insertion
Ethernet Remote Control
• Gigabit Ethernet supports included TCP/IP remote control application; HTTP server included for get/
set control of all parameters.
Front Panel Controls and Indicators
• Rotary navigation cluster, Graphical LCD display, headphone volume control
• 6.3mm front panel headphone connector, +12dBu Max into 600 Ohms
DTV AUDIO PERFECTION FROM PRODUCTION TO TRANSMISSION
TELOSALLIANCE.COM
LINEAR ACOUSTIC | AERO.10
Serial Metadata
• 9-pin female D connector; 115.2 kbps; pinout per SMPTE 207M (RS-485); Designed to directly
interface with Dolby serial metadata (SMPTE RDD6)
Power Requirements
• Dual power supplies, each rated at 100-264 VAC, 50/60Hz, auto-sensing, 100W max. total
Dimensions and Weight
• One rack unit- 1.75”H x 19”W x 15.5”D (44 x 483 x 394 mm) Net weight: 9 lbs (4 kg); shipping: 12 lbs
(5.4 kg) approximate.
Environmental
• Fan cooled. Operating: 0 to 50 degrees C, non-operating –20 to 70 degrees C.
Regulatory
North America: FCC and CE tested and compliant, power supply is UL approved.
Europe: Complies with the European Union Directive 2002/95/EC on the restriction of the use of certain
hazardous substances in electrical and electronic equipment (RoHS), as amended by Commission
Decisions 2005/618/EC, 2005/717/ EC, 2005/747/EC (RoHS Directive), and WEEE.
Warranty
Standard 2-year limited parts and labor
DTV AUDIO PERFECTION FROM PRODUCTION TO TRANSMISSION
TELOSALLIANCE.COM
LINEAR ACOUSTIC | AERO.SOFT
AERO.soft™
Enterprise-wide Audio Processing
OVERVIEW
AERO.soft is enterprise-wide audio and loudness management software for high-density applications
featuring the trusted AEROMAX® adaptive wideband and multiband, multistage, ITU-compliant,
loudness control algorithm. An advanced ITU Limiter (AI) and ITU BS. 1770-3 metering guarantees fast,
accurate and measureable compliance with ATSC and EBU R-128 loudness recommended practices
while maintaining the highest possible audio quality. AERO.soft is also ready for use with the patented
Linear Acoustic Intelligent Dynamics® and Carbon hybrid metadata processing,
Upmixing is provided by the Hollywood-approved UPMAX® II algorithm which delivers engaging
5.1-channel audio from two-channel sources. AutoMAX™ detection and switching between 2.0 and 5.1
surround sources makes full time 5.1 surround output easy to achieve with viewer-pleasing results.
AERO.soft runs on the new Linear Acoustic AERO.soft Processing Engine hardware. Each Processing
Engine is capable of handling up to eight AMX processing instances and 16 stereo pairs of AES67/
Livewire I/O. The eight AMX instances can be any combination of AMX5.1 (5.1+2+2), AMX2.0 (2+2+2) or
AMX5x2 (2+2+2+2+2). All AMX instances support SAP/DVS, LoRo/LtRt downmix, local audio insertion,
and optional Audio Description (warble tone) functionality. ITU-R BS.1770-3 or EBU R128 meters are
present on each audio program output. Linear Acoustic Crowd Control™, which eliminates viewer
complaints about “missing” or “hard to hear” dialogue, is now available for both 2.0- and 5.1-channel
sources.
DTV AUDIO PERFECTION FROM PRODUCTION TO TRANSMISSION
TELOSALLIANCE.COM
LINEAR ACOUSTIC | AERO.SOFT
Dolby® Digital, Dolby Digital Plus, and Dolby E decoding and encoding are optionally available for each
AMX instance as is Nielsen® watermark encoding for AMX5.1 and AMX2.0 instances.
An included comprehensive TCP/IP remote control application provides control over all system settings,
processing and Dolby Digital coding parameters, per-channel signal presence, and loudness metering
from any Windows PC. Multiple logs of LKFS loudness values, True Peak, and Dolby dialnorm and acmod
values are provided for each program output. The remote control application also delivers remote audio
monitoring so the user can audition signal quality anywhere link bandwidth permits. An http server is
also included for log retrieval and provides control of all parameters and retrieval of status via network
commands.
Audio over IP (AoIP) connectivity via Livewire+™ AES67 enables enterprise-wide audio access. Audio from
any video source can be put on the AoIP network using a wide variety of Livewire+™ AES67 compliant
devices available from Linear Acoustic, Axia Audio, and other 3rd party vendors.
FEATURES
• Very high channel capacity software runs real-time on AERO.soft Processing Engine
• I/O independence via Livewire+™ AES67 Audio over IP
• Linear Acoustic AEROMAX loudness/dynamics control
• UPMAX II automatic upmixing and downmixing with AutoMAX 5.1 surround/2.0 detection
and switching
• New Advanced ITU (AI) Limiter
• AMX5.1, AMX2.0 and AMX5x2 Instances available
• ITU-R BS.1770-3 compliant metering
• Extensive remote GUI over IP control enables metering, control, and local audio monitoring (up to 5.1
channels)
• Internal logging of all program outputs: 24 hour, 48 hour, 7 day logs of 3s, 10s, 30s integration, TP and
DN, and 1 hour/100ms logs
• Dolby Digital, Dolby Digital Plus, and Dolby E encoding and decoding options
• Nielsen watermark encoder option
DTV AUDIO PERFECTION FROM PRODUCTION TO TRANSMISSION
TELOSALLIANCE.COM
LINEAR ACOUSTIC | AERO.SOFT
IN DEPTH
Livewire®
Since 2003, broadcast facilities have been using Axia Livewire AoIP (audio over IP) technology to route
hundreds of audio channels using standard Ethernet switches, cables, and other components. With
thousands of devices on the air every day, reliability is proven.
Livewire is lightning fast, simple to implement, easy to manage, and now as Livewire+™ AES67, fully
compliant with the AES67 standard and ready to incorporate new standards as they are developed. It
significantly reduces the number of cables and wires needed to transport and share audio throughout
the broadcast plant.
Nearly any audio source – analog or AES, and now SDI – and myriad equipment from phone systems to
codecs to satellite receivers – can be a part of the AoIP network.
SPECIFICATIONS
Processing
• AEROMAX multistage adaptive wideband and multiband loudness and dynamic range control with
ITU-R BS.1770-3 loudness metering
• UPMAX II two-channel to 5.1 channel upmixing and downmixing, automatically bypasses discrete
content
• Includes extensive support for SAP/DVS, voiceover, downmix auto replacement of SAP/DVS and
optional Audio Description (warble tone) functionality
• 16 Livewire+™ AES67 pairs of I/O
• Linear Acoustic Intelligent Dynamics works with Dolby Evolution as part of the Intelligent Media
Platform
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LINEAR ACOUSTIC | AERO.SOFT
Remote Control
• Dedicated TCP/IP remote control application provides extensive metering, control, system
management and remote monitoring of one or many instances; http server included for log access and
also for control of all parameters using commands over the network
Options
• Any combination of eight AMX instances:
• AEROMAX AMX5.1 instances with dual UPMAX II upmix/downmix
• AEROMAX AMX 2.0
• AEROMAX AMX5x2
• Up to 8 Dolby Digital, Dolby Digital Plus, or Dolby E encoders or decoders (one per AMX instance)
• Up to 8 Audio Description Engines (warble tone) (one per AMX instance)
• Up to 8 Nielsen watermark encoders (one per AMX Instance)
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TELOS ALLIANCE | SDI XNODE
Telos Alliance® SDI xNode™
Livewire+™ AES67 AoIP Interface
OVERVIEW
SDI xNode is a compact and powerful HD/SD-SDI to Livewire+™ AES67 Audio over IP interface. It
provides two independent HD/SD SDI inputs and outputs with audio de-embedding and re-embedding
for up to eight audio pairs.
Video delay (up to1000ms) for each SDI output ensures audio/video synchronization can be
maintained. Any embedded audio pair(s) can be shuffled between the SDI input and output and
Livewire+™ AES67 input audio pairs can be re-embedded to any SDI audio output pair, all in a ½ width
1RU rack space form factor.
Standard xNode front panel and web browser interface controls are provided. An auto-ranging internal
power supply is standard and an external redundant supply is optional.
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TELOS ALLIANCE | SDI XNODE
FEATURES
• Two relay-bypassed 3Gb/s SDI inputs with access to all audio channels
• Dual, independent compensating video delays
• Up to 16 channels of audio via Livewire+™ AES67
• Includes all standard xNode controls including front panel and web interface
IN DEPTH
Livewire
Since 2003, broadcast facilities have been using Axia Livewire AoIP (audio over IP) technology to route
hundreds of audio channels using standard Ethernet switches, cables, and other components. With
thousands of devices on the air every day, reliability is proven.
Livewire is lightning fast, simple to implement, easy to manage, and now as Livewire+™ AES67, fully
compliant with the AES67 standard and ready to incorporate new standards as they are developed.
It also significantly reduces the number of cables and wires needed to transport and share audio
throughout the entire broadcast plant.
Nearly Any audio source – analog or AES, and now SDI – and myriad equipment from phone systems to
codecs to satellite receivers – can be a part of the Livewire network.
SPECIFICATIONS
HD/SD-SDI I/O
• Auto-sensing 3GHz HD/SD-SDI with de-embedding of up to 16 channels total; de-embedded audio
can be routed to Livewire output and/or re-embedded to the SDI output with SMPTE 292M (HD-SDI)
and SMPTE 259M (SD-SDI) support
Livewire+™ AES67
• Fully AES67-Compliant, with Livewire+ AES67
Reference
• Livewire at audio input, SDI-delivered audio clock at output. SDI clock can be provided to Livewire
network if desired.
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TELOS ALLIANCE | SDI XNODE
Ethernet
• Two RJ-45 100BASE-T connections available for management or AoIP interface.
Power Requirements and Consumption
• 95-240 VAC, 50/60 Hz, 15W maximum. Redundant power sourcing available via available external
+12VDC input.
Dimensions and Weight
• 1RU high x 1/2RU width - 1.72”H x 8.5”W x 11.75”D
• (44 x 216 x 298mm) Net weight: 7 lbs. (3.2 kg), approximate.
Shipping Dimensions and Weight
• 12”W x 16”D x 8”H (305 x 406 x 203mm)
• Net weight: 11 lbs. (5 kg), approximate.
Regulatory
North America: FCC and CE tested and compliant, power supply is UL approved.
Europe: Complies with the European Union Directive 2002/95/EC on the restriction of the use of certain
hazardous substances in electrical and electronic equipment (RoHS), as amended by Commission
Decisions 2005/618/EC, 2005/717/ EC, 2005/747/EC (RoHS Directive), and WEEE.
Warranty
Standard 2-year limited parts and labor
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LINEAR ACOUSTIC | LQ-1000
LQ-1000™
Loudness Quality Monitor
OVERVIEW
LQ-1000 provides simultaneous loudness metering per ITU-R BS.1770-3 and EBU R128 standards
(including True Peak readings) for two independent programs (5.1+2) as well as the downmix of the
primary input.
Loudness Loudness logs can be accessed via the built-in HTTP server including logs for the past 24
hours, 48 hours, 7 days, a user-controlled time period, and a brand-new 70 minute log that stores ITU-R
BS.1770-2 100ms measurements every 100ms.
LQ-1000 now utilizes the same NfRemote application already used by AERO.10™, AERO.100™, and
AERO.2000™ for remote connectivity making configuration and monitoring easier and more flexible
than ever. Multiple users can connect to a single LQ-1000 simultaneously. Remote recall of input
configurations and control via IP commands is also supported.
I/O includes AES and auto-sensing HD/SD-SDI. In addition to accepting PCM signals, Dolby Digital (AC-3),
Dolby Digital Plus, and Dolby E decoding are now standard as are dual internal redundant power supplies
and a VGA output to allow the use of an external monitor or incorporation into a multi-viewer array.
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LINEAR ACOUSTIC | LQ-1000
FEATURES
• LKFS metering for two programs – 5.1+2
• ITU-R BS.1770-3 and EBU R128 compliant
• HD/SD-SDI I/O
• AES I/O
• Dolby Digital (AC-3), Dolby Digital Plus, and Dolby E decoding standard
• Large color-coded numeric display shows current loudness value
• Three additional bar-graph meters with adjustable integration times
• Histogram shows loudness history and trends
• Front-panel buttons for Start, Stop, and Reset
• Logging to USB drive or network for all meters and programs
• Built-in HTTP server for retrieval of loudness logs
• Front panel headphone output
• GPIO for external start/stop/reset and alarms
• Dual PSU
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LINEAR ACOUSTIC | LQ-1000
IN DEPTH
Color Display
The bright LCD front panel display presents a wealth of important loudness information all at once,
but remains easy to read at a glance. Critical loudness parameters like short, medium, and long term
loudness, loudness history, current peak level, maximum peak level, and the loudness target are
displayed.
Loudness Speedometer™
The most important information – the current LKFS loudness value – is boldly displayed as a numeric
value and is color-coded to represent the roughly 16dB-wide loudness “comfort zone” which is aligned
around the adjustable target level. The visual is simple: blue is too quiet, green is just right, yellow is
getting loud, and red is too loud.
Network Logging
Readings from each of the four meters for each program (5.1-channel and 2-channel) can be saved to a
USB drive or external network drive. Loudness data for the past 24 hours, 48 hours, 7 days, plus a userdefined period of time is stored in the efficient and universal .csv format and can easily be retrieved from
the built-in HTTP server.
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LINEAR ACOUSTIC | LQ-1000
SPECIFICATIONS
Metering Standards
• ITU-R BS.1770-3 and EBU R128
Audio Input and Decoding
• Baseband input and Dolby Digital (AC-3) decoding standard
• Available Dolby Digital Plus and Dolby E decoding
AES I/O
• Four 75-Ohm AES inputs and outputs via female BNC connectors; Outputs of selected inputs: AES,
De-embedded SDI, Dolby decoded. Signal levels per SMPTE 276/AES-3ID-2001
HD/SD-SDI I/O
• Auto-sensing HD/SD-SDI (SMPTE 292M/259M) inputs up to 1080i/60/59.94/50Hz, access to all 16
audio channels plus VANC metadata per SMPTE 2020M methods A and B.
Re-clocked HD/SD-SDI output
Headphone Output
• I/4’’ (6.35mm) front panel connector with volume control
Parallel GPI/O Control Port
• 25-pin female D connector, 0-5V TTL levels for external start/stop/reset and alarms
Serial Metadata Input
• 9-pin female D connector, 115 kbps per SMPTE 207M (RS-422/485): Directly interfaces with Dolby
metadata (SMPTE RDD6)
VGA Output
• 640x480 for connection to external monitor or multi-viewer
Ethernet
• Gigabit Ethernet via RJ45 for HTTP access and network logging.
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LINEAR ACOUSTIC | LQ-1000
Remote Control
• As of 2016, Windows®-compatible TCP/IP remote control with included application for full setup and
control, ITU-R BS.1770 metering for all programs, encoder statistics, and return audio for remote
monitoring (network speed permitting). HTTP server allows get/set control from PC.
Power Requirements
• Dual power supplies, each rated at 100-240 VAC, auto-ranging, 100 W maximum
Dimensions and Weight
• 3.5”H (2RU) x 19”W x 17”D; (89 x 483 x 432mm)
• Net weight 10.8 lbs. (4.9 kg), approximate.
Shipping Weight and Dimensions
• 22”W x 20”D x 9”H (559 x 508 x 229 mm)
• Net weight:16 lbs. (7.3 kg), approximate.
Regulatory
North America: FCC and CE tested and compliant, power supply is UL approved.
Europe: Complies with the European Union Directive 2002/95/EC on the restriction of the use of certain
hazardous substances in electrical and electronic equipment (RoHS), as amended by Commission
Decisions 2005/618/EC, 2005/717/ EC, 2005/747/EC (RoHS Directive), and WEEE.
Warranty
Standard 2-year limited parts and labor
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LINEAR ACOUSTIC | MT2000
MT2000™
Multichannel Bitstream Analyzer
OVERVIEW
MT2000 is a portable bitstream analyzer and signal generator which includes Dolby® E, Dolby Digital
(AC-3), and Dolby Digital Plus decoding and test signal generation.
The internal ITU-R BS.1770 loudness meter supports ATSC and EBU measurements and includes
selectable Dolby Dialogue Intelligence™ gating for accurate, unattended measurement of long-form
dynamic content.
AES, MADI, HD/SD-SDI, and TOSLINK optical I/O are standard. HDMI input is optional. An external
auto-ranging DC power supply/charger, reference video adapter, USB metadata adapter, and road-ready
custom Mil-Standard high-strength hard case are included.
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LINEAR ACOUSTIC | MT2000
FEATURES
• ITU-R BS.1770-1/2/3 loudness metering
• Built-in test-signal generator
• Accepts signals via MADI, AES, TOSLINK optical, or HD/SD-SDI connectors
• Optional HDMI and DVB-ASI inputs
• Decodes and outputs Dolby Digital/Plus/E and PCM bitsteams
• Bright OLED display
• Built-in monitor speaker
• Powered by an internal NiMH rechargeable battery pack or from its DC power port via included
universal power supply
• New for 2016: Onboard SDI video signal generator with selectable output resolution, streamlined
menu structure, and 16-channel metering view. Existing units can be updated in the field via software
download.
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LINEAR ACOUSTIC | MT2000
IN DEPTH
The MT2000 accepts signals via MADI/AES, TOSLINK™ optical, and HD/SD-SDI connectors. Optional
DVB-ASI and HDMI signal inputs are available. The unit identifies the format of the selected input signal
and activates the appropriate built-in decoder. Monitoring capabilities include error detection at the
AES3 layer and within the coded audio layers, including SMPTE 337 formatting information and Dolby E
guard band position.
In addition to displaying audio signal statistics and metadata, the MT2000 includes ITU-R BS.17701/2/3 loudness measurement with selectable Dolby Dialogue Intelligence™ to support ATSC A/85 and
EBU R128.
An extensive set of useful Dolby Digital, Dolby Digital Plus, and Dolby E test bitstreams is stored
internally, and users can modify the set in the field via software download. The MT2000 can generate
the selected bitstreams simultaneously on all output connectors, even while receiving and decoding an
input signal. The MT2000 is also capable of generating two-channel PCM signals. In this mode, the user
can select the output waveform type (white noise, pink noise, sine, square), amplitude, and frequency.
Test signals and analysis are also provided for latency and basic lip sync.
Signals are provided simultaneously via the MADI/AES and TOSLINK optical outputs and can be reembedded into any of the SDI pairs. Output can be the original input signal, a multichannel PCM decoded
version of the input signal, test signals, or, in the case of the SDI output, a combination of all of these.
Inputs can be used as sources for embedding even if not used for decoding thus channel shuffling can be
easily accomplished.
A bright yellow OLED display and integrated rotary navigation cluster provide straightforward menu
navigation and function adjustement. A standard 1/8-inch stereo headphone jack can be switched to
monitor any two decoded channels or a downmix of the whole program. An integrated monitor speaker
provides a surprisingly loud output useful for quick checks and for emulating sound systems found in
portable devices.
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LINEAR ACOUSTIC | MT2000
SPECIFICATIONS
Audio Formats
• Dolby Digital, Dolby Digital Plus, and Dolby E inputs and outputs of test bitstreams; Stereo (AES/
TOSLINK optical) and multichannel PCM (SDI and MADI) input; Live generation of PCM waveforms such
as white noise, pink noise, sine, and square waves, latency test signal, A/V sync test pulse (for beep/
flash).
Metering
• ITU-R BS.1770 loudness measurement with Dolby Dialogue Intelligence; Error detection at the AES3
layer and within the coded audio layers, including SMPTE 337 formatting information and Dolby E
guard band position.
AES and MADI I/O
• All connectors are 75-Ohm BNC female; Main inputs with 75-Ohm internal termination; Signal levels
per SMPTE 276M/AES-3ID-2001. Compatible with consumer S/PDIF connections. AES I/O connectors
also serve as I/O for MADI.
HD/SD-SDI I/O
• Auto-sensing 3GHz HD/SD-SDI (SMPTE 292M/259M/424M), up to 1080p/60/59.94/50Hz, access to
audio and VANC metadata and timecode.
•
TOSLINK Optical I/O
• Supports consumer IEC 61937 input and output.
DVB-ASI (Option)
• DVB-ASI (ETSI TR 101 891 v1.1.1) Transport Stream Input (via SDI connector). Audio PID can be
selected for decoding and measurement.
HDMI (Option)
• Supports monitoring of embedded audio bitstreams from set-top boxes and other consumer devices
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LINEAR ACOUSTIC | MT2000
Navigation
• Front Panel Controls and Indicators
• Rotary joystick navigation cluster plus graphical OLED display
Ethernet
• 10/100BT via RJ45 for future applications
USB
• Connection for software updates and future applications.
Metadata Input
• VANC from SDI input, or embedded metadata from other inputs
Video Reference Input
• Dedicated VRef input for Dolby E bitstreams
Headphone Output
• 3.5mm (1/8-inch) side connector, +12 dBu max into 600-Ohms
Power Requirements
• Internal NiMH rechargeable battery; Power and charge via dedicated DC input
Dimensions and Weight
• 7.9”H x 4”W x 1.6”D (200 x 100 x 41 mm)
• Net weight: 3 lbs (1.36 kg), approximate.
Shipping Weight
• 6 lbs. (2.7 kg), approximate.
Environmental
• Fan cooled. Operating: 0 to 50 degrees C, non-operating –20 to 70 degrees C.
DTV AUDIO PERFECTION FROM PRODUCTION TO TRANSMISSION
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LINEAR ACOUSTIC | MT2000
Regulatory
North America: FCC and CE tested and compliant, power supply is UL approved.
Europe: Complies with the European Union Directive 2002/95/EC on the restriction of the use of certain
hazardous substances in electrical and electronic equipment (RoHS), as amended by Commission
Decisions 2005/618/EC, 2005/717/ EC, 2005/747/EC (RoHS Directive), and WEEE.
Warranty
Standard 2-year limited parts and labor. 90 days for battery.
Supplied Accessories
Carrying case, universal DC power supply, video reference input cable, USB memory stick
containing user manual.
DTV AUDIO PERFECTION FROM PRODUCTION TO TRANSMISSION
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MINNETONKA | AUDIOTOOLS SERVER
AudioTools™ Server
Quality-First Audio Automation
OVERVIEW
AudioTools Server is a collection of enterprise-ready audio solutions designed to add file-based expertise
to automated environments. The AudioTools Server family represents decades of expertise encapsulated
in flexible, focused packages of audio specialization.
AudioTools Server automates the most sophisticated audio tasks and offers a wide variety of processing,
specifically created for use in cable, satellite, terrestrial and IPTV, radio, and post production facilities.
As a pure software platform running on commodity hardware, AudioTools Server is flexible and
customizable, allowing for new workflows as requirements inevitably change.
• State-of-the-art audio processing
• Customized, efficient file-based workflows
• Unrivaled loudness tools
• Compliance to broadcast standards
• Interoperability with major environments
• Modular scalable platform
QUALITY-FIRST AUDIO AUTOMATION | CERTIFIED CODECS FOR WORKSTATIONS, ENTERPRISE & OEM
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MINNETONKA | AUDIOTOOLS SERVER
AudioTools Server - Leader in Loudness
Loudness Control is a significant use case that employs Linear Acoustic APTO™ loudness
processing. Our loudness normalization processes are designed to fully preserve the existing audio
and only apply a gain change combined with optional peak limiting. There are other use cases that
require changing the dynamic range and other more complex parameters of the audio content
and can for example look after dialog intelligibility. AudioTools Movie Adaptation Automation is a
collection of intelligent loudness profiles, adapting a theatrical audio mix for broadcast, and creating
the best possible audio experience for all modern platforms such as OTT/web, mobile/handheld and
VOD/SVOD.
FEATURES
Use Cases
Loudness Measurement & Adjustment
• State-of-the-art loudness control based on international standards & practices. AudioTools Movie
Adaptation Automation profiles for improved and compliant dialogue intelligibility in high-dynamic
range content.
Dolby Automation
• Automated Dolby encoding and decoding including metadata handling. Dolby E quality control, with
optional correction.
Audio Adaptation
• Automated adaptation of audio content to specific output specifications, including upmix, downmix,
channel management …
Quality Control
• Audio specific quality control of pure audio or container formats, including channel assignment
detection and correlation check.
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MINNETONKA | AUDIOTOOLS SERVER
IN-DEPTH
The Server Architecture
The AudioTools Server communicates directly with modules to configure the overall system for the
required functionality. AudioTools Server, as a system, can then interface directly with your existing DAM,
CMS or DBM workflow manager.
Modular Design
While a modular architecture allows AudioTools Server to always be “state-of-the-art”, the real power
of this approach is using workflows to combine module functions in a sequential or highly conditional
profile. AudioTools Server is also a VST plug-in host, allowing for 3rd party plug-ins to be used as part of
an overall AudioTools Server workflow.
Enhanced Workflows with AudioTools Server
AudioTools Workflow Control is the command and control option for AudioTools Server that enables
standalone operation along with support for threaded multiple concurrent processes, load balancing and
dynamic reconfiguration of workflows on the fly. AudioTools Server can deploy floating licenses through
a license server, offering a scalable system for small businesses or enterprise class facilities.
AudioTools Server is internally driven from tailored XML profiles. For an operator, preset workflows can
easily be called up through the AudioTools OPERATOR App, while Queue Control provides an overview
with detailed access to all running processes.
AudioTools OPERATOR
Operators get assistance and access on parameters through a simple, custom user interface.
Example implementations:
Advanced Loudness Adaptation (left) and Dolby E encoder (right)
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MINNETONKA | AUDIOTOOLS SERVER
Integration
AudioTools Server supports manual job submission, hot folders, and full web services integration,
guaranteeing that AudioTools Server can fit into any environment and on any level the user requires.
Integration Partners
• Arvato
• Aspera
• Aveco
• AVID Interplay MAM
• Civolution
• Dalet & AmberFin
• Dolby
• Evertz Mediator
• Geminisoft
• Harmonic
• IBM
• Root 6
• Sony
• Tedial
• Telestream
• Vector 3
• VIZRT
• Wohler RadiantGrid
SPECIFICATIONS
Content & Formats
AudioTools Server is a complete solution for managing and processing audio formats. With audio at the
center of a complex audio ecosystem, a payload may contain “containerized” video assets, transport
streams, Dolby–encoded content, and metadata, requiring that each layer of the asset must be carefully
handled.
Formats:
• Audio: Linear PCM, AIFF, WAV
• Codecs: Dolby E, Dolby Digital, Dolby Digital Plus, Dolby Pro Logic II, mp2, mp3, MPEG-4, HE-AAC, AAC
• Container: MXF, QuickTime™, LXF, GXF, Transport Streams, Program Streams
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MINNETONKA | AUDIOTOOLS CLOUD
AudioTools™ Cloud
Quality-First On-Demand Audio Processing
OVERVIEW
AudioTools Cloud™ is an advanced audio processing solution for audio, video, and broadcast
professionals that provides loudness control, encoding, decoding, channel management, frame rate
conversion, quality control, and container management from an easy-to-use user interface designed for
the Amazon AWS Marketplace.
AudioTools Cloud is based on Minnetonka’s AudioTools™ Server – a platform that delivers interoperable,
scalable, file-based audio automation. AudioTools Server has become the number one enterprise level
platform for automated and unattended file-based audio processing and has helped broadcasters add
Loudness Management processes to their existing video-centric file-based environments.
From the occasional job, to expanding throughput for higher volume workloads, AudioTools Cloud brings
these proven processes to the cloud, allowing on-demand, case-by-case use (OPEX) vs larger CAPEX
sized projects. Businesses only pay for the infrastructure they need, when they need it.
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MINNETONKA | AUDIOTOOLS CLOUD
FEATURES
• Professional file-based audio processing
• Extensive selection of job types
• Intuitive user interface enables complete control over every job parameter
• Support for global loudness compliance standards
• SurCode encoding and decoding technologies included
• 3 flexible configurations
Submitting Jobs to AudioTools Cloud
AudioTools Cloud ON-DEMAND is available as different Amazon EC2 instances. Each instance will launch
the AudioTools Cloud Web Client Interface. Users can choose to upload content to attached storage EBS
(Elastic Block Store) or Amazon S3 (Simple Storage Service). After the upload is complete, select your
input buckets, output buckets, and audio processing parameters.
Users pay for the EC2 instance, storage, and AudioTools Cloud software.
IN-DEPTH
AudioTools Cloud ON-DEMAND
AudioTools Cloud ON-DEMAND is designed for the Amazon AWS Market Place and offers ready-to-use
audio processing profiles. A simple click-through browser based “configurator” is used to assign input
and output locations and file types, or adjust loudness target levels for different specifications. These
configurations can be downloaded and stored locally as templates for future use.
Self-service: Users choose what they want and when they want it.
Scalable: Users can choose how much capacity they want to ramp up if necessary.
AudioTools Cloud BYOL - Bring Your Own License
AudioTools Server users can benefit from a cloud based deployment by adding AudioTools Server
instances to any public or private cloud environment. An AudioTools LicenseServer will allow for floating
licenses across all AudioTools Server instances in the cloud, on premise or any VM or datacenter
deployment.
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MINNETONKA | AUDIOTOOLS CLOUD
The AudioTools Server system in the cloud will contact the LicenseServer for available licenses,
benefiting from cloud based flexible scalability within the existing license pool. AudioTools Cloud BYOL
is perfect for users that want the flexible scalability of cloud based processing, combined with defined
processing profiles and licenses. This is also the ideal strategy for smoothly and gradually moving work
from an on premise installation to cloud based processes.
AudioTools Cloud Cloud Node
AudioTools Server v4 will offer an on-demand Amazon Cloud-based AudioTools Server instance in
addition to an existing AudioTools Server installation. The cloud based service is being added as a
processing node for AudioTools Server. AudioTools Workflow Control can then use the Cloud Node to add
more processing resources to a local system. In a load-balanced environment, if a local system is not
licensed for a specific ATS module, Workflow Control can assign those tasks to the Cloud Node, which by
default includes all possible modules and licenses for AudioTools Server. AudioTools Server Cloud Node
is the perfect add-on for flexible scalability and additional licenses on a project-by-project basis.
SPECIFICATIONS
AUDIO PROCESSING OPTIONS
Frame Rate Conversion
• Film - NTSC - PAL
• Pitch Shift
• Time Stretch
• Sample Rate Conversion
Loudness Control
• EBU R 128 Loudness Adjustment
• CALM A/85 Loudness Adjustment
• AS-11 UK DPP
• Loudness Measurement
• Linear Acoustic APTO audio processing
• Advanced Loudness Adaptation
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MINNETONKA | AUDIOTOOLS CLOUD
Quality Control
• Dolby E Metadata Verification
• Dolby E Wave File Analyzer
• Channel Detection
• Program Correlation Check
• Silence Detection
• Data Corruption Detection
Audio Codecs
• Dolby Digital (Plus) Encoding and Decoding
• Dolby E Encoding and Decoding
• DTS-HD Encoding and Decoding
• MP2/MP3 Encoding and Decoding
• Transcoding Dolby E to Dolby Digital (Plus) streams
• AAC Encoding and Decoding
Container Management
• Extract audio from MXF, LXF, GXF, QuickTime™, and Transport Streams
• Re-wrap audio to MXF, LXF, GXF, QuickTime™, and Transport Streams
Channel Management
• Upmix with Linear Acoustic UPMAXTM
• Downmix
• Channel Swapping
• Channel Replacement
• Channel Mixing
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MINNETONKA | AUDIOTOOLS FOCUS
AudioTools™ FOCUS
for Loudness Control
OVERVIEW
Loudness in a Video World
AudioTools FOCUS for Loudness Control is an easy to use, standalone Windows application that takes
the complexity out of audio loudness processing work. Proven AudioTools presets conforming to
every loudness standard are built in and ready to use or modify. With all the standards, file formats
and deliverables needed today, loudness management adds one more layer of intricacy to an already
complex ecosystem. In a video– centric world, we focus on handling the complexity and dependencies of
automating loudness …AudioTools FOCUS brings audio control into focus.
FEATURES
AudioTools FOCUS for Loudness Control is intended for:
• Sole proprietors needing painless yet high quality logging & adjustment
• Post houses & networks needing container handling
• Any facility that doesn’t have deep automation in place
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MINNETONKA | AUDIOTOOLS FOCUS
Hit All Targets
AudioTools FOCUS for Loudness Control makes quality loudness management painless, taking the
complexity of files, containers and processing, and making it all available in an easy to use, preset–driven
application. A unique AudioTools feature is its ability to hit all loudness targets, not just one or two. With
AudioTools FOCUS’ logic–driven approach, you can simultaneously comply with Program Loudness,
Loudness Range, and True–Peak, without exceeding your Maximum Momentary or Maximum Short
Term loudness thresholds.
Intricacy Unraveled
Simply type in your target parameter changes to the factory presets, and let FOCUS do the rest:
• Start with a preset and type in your target requirements for each delivery spec
• Define your program configuration
• Set your file I/O - Watched Folders or Files
• Hit Go!
AudioTools FOCUS provides:
• Loudness management with quality-first results
• Complete compliance with North American (CALM) & international loudness control standards
including: ITU BS.1770, 1770-2/3, EBU R 128, ARIB TR-B32 & OP-59
• Hot Folder or File-driven processing
• MXF & QuickTime container handling – extract/process/rewrap
• Loudness adjustment or pure logging
• File type auto-detection
• Clearly labeled, proven presets right out of the box
IN-DEPTH
AudioTools FOCUS – An Affordable, Quality-First Solution For
Loudness Control
Not only are there many variations of video file formats, but a quality loudness management process
itself is complex. AudioTools FOCUS for Loudness Control provides:
• Programme Loudness adjustments with combined True–Peak limiting
• Programme Loudness combined with Maximum Short Term Loudness control
for short for & interstitials
• Active & passive” Max. Short Term Loudness combined with Max Momentary limiting
• Pure gain adjustment versus dynamic processing
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File-based In Mind
The world is moving toward IT dominated, file–based workflows, and audio is only a part of the overall
video picture. AudioTools FOCUS extracts the audio, processes that essence, preserves any audio
metadata and rewraps the audio back into the container without modifying the video. This streamlines
your workflow while ensuring downstream compatibility.
The ITU-R BS.1770 Programme Loudness standard was defined as file-based. AudioTools FOCUS
directly answers that need with a standalone, software solution for loudness control automation.
Simplicity…It’s In The Details
Unlike hardware investments that cannot keep up with changing business strategy, delivery/platform
requirements and government mandates, AudioTools FOCUS for Loudness Control is a pure software solution.
SPECIFICATIONS
The AudioTools FOCUS and Server Family
Features of
AudioTools ...
Operation
Automation through watch folders
FOCUS
FOCUS Server
FOCUS
Server Plus
Server




Automation through web services API



Number of concurrent jobs
1
4
4
flexible
licensing
Presets via graphical UI










Configurable graphical UI

SOAP integration & XML scriptable
Custom Workflows
Format Support

LPCM Audio: WAV, BWF, AIFF

Compressed Audio: MP2, MP3, AAC, Dolby
Digital Plus
Audio: Dolby E

Container: MXF, QuickTime


8
16
Video: Transport Stream, GXF
Audio channels
Loudness Processing





16
128
Audio program configurations
Presets
Presets
Presets
Scriptable
ITU 1770 & EBU R 128 Measurement




Program Loudness Adjustment




Momentary & Short-Term Loudness
Adjustment




True-Peak Limiting




Loudness Range Adjustment




Dialog-based level adjustment




optional
optional
optional
optional
optional
mandatory
NEW
AudioTools Advanced Loudness
Adaptation
AudioCare
Service Level Agreement (per year)
optional
AudioTools FOCUS and FOCUS Server can be upgraded to a full AudioTools Server configuration, with 80%
of your AudioTools FOCUS purchase applied as a credit.
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MINNETONKA | AUDIOTOOLS LOUDNESS CONTROL FOR HARMONIC PROMEDIA™ CARBON
AudioTools™ - Loudness Control for
Harmonic ProMedia™ Carbon
OVERVIEW
The AudioTools Loudness Control for Harmonic Pro-Media™ Carbon plug-in adds versatile and
convenient loudness control to Harmonic’s file-based ProMedia™ Carbon transcoding software. The
plug-in provides comprehensive loudness measurement and flexible loudness adjustment using the
same technology that powers Minnetonka Audio’s AudioTools Server - the worldwide benchmark for
file–based loudness control.
FEATURES
All Standards Supported
ALCR is compliant with all current standards, delivers your choice of loudness adjustment actions, and
covers all essential metrics - quantifying Programme Loudness along with Maximum True Peak Level.
EBU and CALM–specific profiles are provided to address all international regulations. Measurement
results are formatted as XML, making them available to ProMedia™ Carbon for logging and subsequent
reporting. The XML results are also compatible with the visualization and reporting application from
Videomenthe and with other XML tools.
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MINNETONKA | AUDIOTOOLS LOUDNESS CONTROL FOR HARMONIC PROMEDIA™ CARBON
More than Stereo
AudioTools Loudness Control for Harmonic ProMedia™ Carbon handles mono, stereo and multichannel
PCM configurations up to 24 channels, including 4 custom program configuration profiles. By processing
all programs in one pass, users save time prepping material, whether it’s a simple stereo or multiprogram configuration.
IN-DEPTH
Recommended Practice
EBU and ATSC recommended practices discourage the use of preset compression and peak limiting for
loudness control in file– based environments. The AudioTools Loudness Control for Harmonic ProMedia™
Carbon plug-in intelligently applies loudness adjustment to the incoming essence to reach the desired
target Program Loudness value. Loudness adjustment can be performed in accordance with EBU R
128 and, as with the measurement module, mono, stereo and multichannel PCM configurations can be
processed with target values set for Program Loudness according to ITU-R BS.1770-3. If the loudness
level deviates from the Target Loudness Value, adjustment is applied, and Maximum True Peak Level can
be limited to previously specified target levels.
Harmonic users now have a comprehensive solution that amplifies the capabilities of ProMedia™ Carbon
— an adaptable AudioTools plug-in to measure and maintain loudness on your already familiar Harmonic
ProMedia™ Carbon platform. AudioTools Loudness Control for Harmonic ProMedia™ Carbon can be
added to an existing video transcoding process, providing loudness control functionality to a large variety
of video and container formats on the fly.
AudioTools Loudness Control for Harmonic ProMedia™ Carbon is also supported in Harmonic WFS
environments allowing users to create machine groups from a collection of authorized nodes.
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SPECIFICATIONS
Measurement
• Supports ITU-R BS.1770-3, EBU R 128 and other international standards & practices
• Is in accordance with the Commercial Advertisement Loudness Mitigation (CALM) Act and ATSC’s
Recommended Practice A/85: Techniques for Establishing and Maintaining Audio Loudness for
Digital Television.
• Measures:
• Programme Loudness
• Maximum Momentary Loudness
• Maximum Short Term Loudness
• Loudness Range
• Maximum True Peak Level
• Sample Peak Level
• Measurement results are formatted as XML in a log file, & for use in 3rd party applications, such as the
Videomethe plotter
• In addition to overall loudness measurement results, maximum overall values are reported as well as
measured loudness parameters throughout the file.
Adjustment
• AudioTools Loudness Control for Harmonic ProMedia™ Carbon applies loudness adjustment to the
incoming essence to reach the desired target Programme Loudness and control the Maximum True Peak.
• Loudness adjustment is performed in accordance with EBU R 128, A/85 & BS.1770-3.
• Mono, stereo and multichannel PCM configurations can be processed - up to 24 channels
• All programs within an asset are processed in a single pass
• Target values can be set for all adjustable parameters.
AudioTools Loudness Control for Harmonic ProMedia™ Carbon can be activated on a single machine
using a license file. A user-supplied USB dongle provides portable licensing. On a single host or node,
AudioTools Loudness Control for Harmonic ProMedia™ Carbon can be used as multiple instances for
parallel processes.
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MINNETONKA | SURECODE FOR DOLBY E
SurCode for Dolby® E
OVERVIEW
The Industry Standard for Dolby E On Your Workstation
Dolby E is the industry standard method of transporting multichannel digital audio across an asset’s
entire post–production life cycle, and SurCode for Dolby E is the industry’s benchmark for file–based
processing of Dolby E. As the first and most complete tool kit for Dolby E, SurCode for Dolby E is a
certified Dolby E encoder and decoder suite that allows fast and easy processing and management of
file-based Dolby E assets and metadata.
• Structured workflow
• Increased productivity
• Flexible license handling
• Encode, decode & monitor
• Broad platform support – Avid, Adobe, Final Cut & VST
• Interoperable with Dolby hardware
• Lowest cost Dolby monitoring solution in the industry
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Easy & Flexible Workflow
SurCode for Dolby E provides a simple and flexible workflow tailored specifically to your platform and
workstation. All platforms, including Pro Tools, Media Composer, NewsCutter, Symphony, VST and Final
Cut are included on a single license.
• Settings are conveniently embedded in your session for instant recall or rework
• Supports all Dolby metadata
• Choose the best way to integrate into your own workflow
Complete Metadata Management
SurCode for Dolby E generates and preserves all Dolby metadata. This gives broadcast professionals the
ability to control channel mode, downmix parameters, profiles, dialog level and other ancillary metadata.
Browse or drag and drop DBMD to read existing metadata since SurCode for Dolby E imports and
exports metadata as:
• DBMD - add your Dolby metadata chunk into WAV headers
• XML - universal format for AudioTools Server or any XML–aware application
• Text - human–readable metadata for manual QC
SurCode for Dolby E Plugin
SurCode for Dolby E includes encoder and decoder plug–ins for Pro Tools, Final Cut, Symphony, Media
Composer, NewsCutter, and VST versions for qualified surround–capable workstations such as Adobe
Premiere Pro CC, Nuendo, Pyramix, Sequoia and Fairlight.
FEATURES
Dolby E Encoder
SurCode for Dolby E Encoder provides in–depth Dolby E encoding, supporting all Dolby E encoder
program configurations. Dolby E and program metadata are displayed and can be updated, all via a
simple, tabbed user interface. The SurCode for Dolby E Encoder plug–in seamlessly integrates with your
workstation using the current session information.
• Pro Tools, Media Composer & NewsCutter AudioSuite; cross–platform VST & Final Cut Pro Export
• Up to 8 input channels, with flexible input routing
• Full metadata management
• Includes plug–in templates & presets
• DBMD import/export
• DP600 emulation option for guard band & other file settings
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• Optionally enable or disable timecode in encoded files
• Frame offset control for tape–based or Dolby hardware layback
• Writes 20, 24 or 32 bit WAV files
• Supports both dual mono or single interleaved WAV files
• Stream metadata from legacy hardware via optional USB-to-RS-422 serial cable
SurCode for Dolby E Decoder
SurCode for Dolby E Decoder decodes Dolby E files or streams, and provides output and routing of audio
streams. Playback control is at your fingertips, with convenient program selection that optionally routes
stereo essence automatically to your default 1/2 bus pair, eliminating the need to repatch. The user
interface displays program configuration, output metering, metadata, decoding status and a decoding
error indicator for both the Dolby E file and individual programs. SurCode for Dolby E Decoder also
enables real time testing and playback for tight and consistent quality control.
• Supports all Dolby E encoder program configurations including 5.1 + 2
• Dolby E Data Bit Depth of 20 or 16 bit
• Up to 8 output channels per file or stream
• Export DBMD chunk, XML or text metadata
• Provides both peak & RMS metering (AAX, VST, RTAS & AudioSuite only)
• Display & update Dolby E + per-program metadata
• Save user–defined presets
SurCode for Dolby E Stream Player
The SurCode for Dolby E Stream Player plug–in delivers real–time decoding capability for all your
post–production stakeholders, even when they are only equipped for stereo routing playback. As a
complement to a complete Dolby E workflow or, a low cost quality control solution for any of your rooms,
SurCode for Dolby E Stream Player easily integrates the Dolby E format into existing workflows across
platforms and applications, providing more choice for video post production professionals. It’s a problem
solver for broadcasters, contractors, second units, location crews and anyone who needs to incorporate
Dolby E into their workflow without the bulk and expense of full surround. SurCode for Dolby E Stream
Player enhances NewsCutter, Media Composer, Symphony, Pro Tools, Final Cut Pro 6 or 7 plus Nuendo,
Pyramix and Fairlight or qualified VST hosts.
• Real–time downmixing of 5.1 to stereo
• Downmix parameters taken directly from Dolby E metadata for a true emulation
• Real–time program selection during playback
• Adjustable downmix headroom control
• Comprehensive Dolby E metadata display
• Export DBMD chunk, XML or text metadata
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IN-DEPTH
SurCode for Dolby E – for Media Composer, Symphony & NewsCutter
For faster–than–realtime processing, the SurCode for Dolby E plug–in appears as an AudioSuite encoder
and decoder, right in the AudioSuite menu of your Avid workstation. For realtime audition from the
timeline, the SurCode for Dolby E Decoder AAX and RTAS plug–ins are also included. The time saving
Mixdown feature for Media Composer, Symphony and NewsCutter allows you to apply the AAX and
RTAS decoders, in a faster–than–real time capacity, to a multichannel bus. It not only decodes the audio
but automatically creates amultichannel track, and prints the decoded PCM audio to that track.
SurCode for Dolby E – for Pro Tools
For faster–than–realtime processing, the SurCode for Dolby E plug–in appears as an AudioSuite encoder
and decoder, right in the AudioSuite menu of your Pro Tools workstation. For realtime audition from
a file, the timeline, or a live input, the SurCode for Dolby E Decoder RTAS and AAX plug–ins are also
included. When only QC or confidence monitoring is required, the low cost SurCode for Dolby E Stream
Player saves money and frees up more costly resources
SurCode for Dolby E - for Adobe Premiere Pro CC
For faster–than–realtime processing, the SurCode for Dolby E Encoder is available as an export plug-in
in the Adobe Media Encoder export engine. SurCode for Dolby E VST Decoder decodes directly from the
Premiere Pro time line. Simply drop a Dolby E encoded file onto the time line and instantiate a SurCode
for Dolby E Decoder in the Audio mixer to either audition or decode to PCM.
SurCode for Dolby E - for Nuendo
For faster–than–realtime processing, the SurCode for Dolby E VST Encoder encodes by selecting the
Audio Mixdown option in the File Export menu. The SurCode for Dolby E Decoder VST plug–in allows
realtime audition from a file, the timeline, or a live input as an Insert plug–in on your multichannel track.
SurCode for Dolby E Stream Player saves you time and resources, decoding in Nuendo in stereo.
SurCode for Dolby E - for Pyramix
For faster–than–realtime processing, the SurCode for Dolby E VST Encoder encodes by selecting the
Project menu, where you select Mix Down. The SurCode for Dolby E Decoder VST plug–in allows realtime
audition from a file, the timeline, or a live input as an Insert plug–in on your multichannel track. For your
B room, SurCode for Dolby E Stream Player saves time and money, with stereo decoding in Pyramix.
SurCode for Dolby E - for Fairlight
For encoding, use the Insert Config menu in the Mixer Megamode. For faster–than–realtime processing,
the SurCode for Dolby E VST Encoder encodes by selecting the Wave menu in the Editor Megamode.
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SurCode for Dolby E - for Final Cut
For faster–than–realtime processing, the SurCode for Dolby E Encoder and Decoder appear as options
in the File Export menu. For confidence monitoring and QC in a stereo environment, SurCode for Dolby E
Stream Player is the lowest cost solution available.
SurCode for Dolby E - for Sequoia
For realtime processing, the SurCode for Dolby E VST Encoder encodes via the Plug-ins menu, where you
select VST FX. The SurCode for Dolby E Decoder VST plug–in gives you realtime auditioning from a file or
the timeline, and SurCode for Dolby E Stream Player lets you monitor Dolby E in stereo right in Sequoia.
SPECIFICATIONS
System Requirements
For specific system requirements and product bundles, visit our web site at www.minnetonkaaudio.com.
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MINNETONKA | SURECODE FOR PRO LOGIC II
SurCode for Dolby® Pro Logic II
OVERVIEW
SurCode for Dolby Pro Logic II – LtRt plug-in for Workstations
SurCode for Dolby® Pro Logic II is a certified Dolby Pro Logic II encoder and decoder that allows fast and
easy processing and management of file-based Dolby Pro Logic assets.
SurCode for Dolby Pro Logic II offers complete Dolby Pro Logic II encoding and Pro Logic II, Pro Logic IIx
and Pro Logic IIz decoding of up to 8 channels of audio. The product enables auditioning, encoding and
decoding of audio, making it easy to produce surround–ready mixes in real time and, being file–based, at
faster than real time. SurCode for Dolby Pro Logic II also decodes back to multichannel audio, making it
the perfect complement to post–production LtRt needs, and multichannel field surround microphones.
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FEATURES
• Complete encoding, decoding, measuring, monitoring and QC for LtRt and LoRo in one unified
environment
• Encode & decode simultaneously in real time to easily optimize your input
• Perfect complement to SurCode for Dolby E Encoder
• v2.5 includes and RTAS plug-in running on Pro Tools 8, 9 HD, 10 or PT 8 LE with Complete Production
Toolkit; VST 32bit, and cross-platform Standalone support.
• V3.0 includes a 32bit AAX plug-in running on Pro Tools 10.3.6 or higher, a 64 bit AAX plug-in running
on Pro Tools 11 & 12 and Media Composer 8.1 or higher, and 64 bit VST for qualified 2.4 surround
capable hosts.
• Built–in loudness measurement for simplified workflows & EBU R 128 & CALM–compliant
deliverables
IN-DEPTH
Built For Loudness Standards
Broadcasters the world over need to deliver loudness–compliant content for mandated delivery
channels, and should correct all material, regardless of channel and platform. SurCode for Dolby Pro
Logic II displays seven different standard loudness measurement parameters:
• ITU 1770 ungated Integrated Loudness in LKFS
• ITU 1770-2/3 gated Integrated Loudness in LKFS
• EBU R 128 Integrated Loudness in LUFS
• Dialog–anchored Integrated Loudness
• Maximum Momentary & Short Term Loudness
• Maximum True–Peak
These real time loudness measurements allow you to adjust and verify objective loudness parameters of
both your source files and encoded LtRt content.
Complete – End to End
SurCode for Dolby Pro Logic II allows an end–to–end, encode/decode cycle for direct control of your
mix’s entire life cycle. In addition to a reference grade LtRt encoder, SurCode for Dolby Pro Logic II also
provides a facility for stereo or LoRo output as well. The encoded LtRt can be monitored as a non–
decoded stereo stream or decoded as a surround stream.
SurCode for Dolby Pro Logic II enables engineers to use SurCode technology in your favorite Avid or VST
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MINNETONKA | SURECODE FOR PRO LOGIC II
DAW, making it a powerful, time-saving tool to efficiently mix directly in your DAW. Now the exact same
phase and amplitude behavior of the Pro Logic II consumer decoding chain can be monitored and levels
corrected if necessary, saving time and ensuring the highest quality output during the mixing process.
SurCode for Dolby Pro Logic II version 3 also offers a cross–platform, standalone version for use in any
workstation environment.
SurCode for Dolby Pro Logic II monitoring and loudness measurement tools and built-in encoding/
decoding features enable mixes to be optimized for all playback situations, regardless of whether the
LtRt is decoded into surround or remains in stereo mode as delivered.
Tools For Today … and Tomorrow
SurCode for Dolby Pro Logic II continues the Minnetonka Audio tradition of supplying discriminating
audio professionals with full featured Dolby codecs. In keeping with our parallel philosophy to streamline
workflows, version 3 integrated loudness measurement saves time and money while reinforcing
industry best practices.
SurCode is the world’s most trusted family of professional codecs for Dolby formats, with pro–level
service and support. As with all members of the SurCode family, SurCode for Dolby Pro Logic II is fully
Dolby–certified, allowing the use of associated Dolby trademarks and branding.
For television, radio, corporate, industrial, games, OTT or Mobile/Hybrid; anywhere your client wants to
deliver engaging, compelling, forward and backwards–compatible content, Dolby Surround is the format
of choice, and SurCode for Dolby Pro Logic II delivers easy LtRt content with complete confidence.
SPECIFICATIONS
System Requirements
SurCode for Dolby Pro Logic II is built for use in Avid’s Pro Tools, Media Composer, NewsCutter and
Symphony, using the AAX, RTAS and AudioSuite formats. The installer provides cross–platform VST
support as well, for Nuendo, Pyramix, Sequoia, Fairlight and other VST 2 compliant hosts. Also installed
are cross–platform, standalone applications for Windows and Mac OS.
For specific system requirements and product bundles, please visit our web site at www.
minnetonkaaudio.com.
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MINNETONKA | SURECODE FOR DOLBY DIGITAL PLUS
SurCode for Dolby® Digital Plus
OVERVIEW
SurCode for Dolby Digital Plus
SurCode for Dolby Digital Plus Encoder converts 5.1, 6.1 and 7.1 surround sound audio to Dolby Digital
Plus (E-AC-3) and Dolby Digital (AC-3) format. The product accepts up to 8 channels of 44.1 or 48 kHz
PCM audio at word lengths of 16, 24 or 32 bits per sample. SurCode for Dolby Digital Plus Encoder
outputs frames through your favorite DAW or to an .ec3/.ac3 or WAV file. It also supports all standard
metadata and pre-processing options.
SurCode – The Standard
Dolby Digital is the industry standard method of delivering discrete surround audio to consumers. As the
most complete toolkit for Dolby Digital and Dolby Digital Plus production, SurCode for Dolby Digital Plus
is the certified encoder and decoder that allows fast and easy processing and management of file-based
linear PCM and Dolby Digital assets. As a plug–in, it runs in all current and legacy workstation formats,
and also includes a cross–platform native version.
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MINNETONKA | SURECODE FOR DOLBY DIGITAL PLUS
FEATURES
SurCode for Dolby Digital Plus offers complete Dolby Digital Plus and Dolby Digital supports up to 8
channels of audio. The product enables auditioning, encoding and decoding of audio, making it easy
to produce discrete surround files in real time and, being file–based, at faster than real time speeds.
SurCode for Dolby Digital Plus also decodes back to multichannel PCM audio, making it the perfect
complement to your post–production needs for film, TV, mobile, gaming and VOD.
• Complete encoding, decoding, monitoring and QC in one unified environment
• Encode & decode simultaneously in real time to easily optimize your metadata
• Encoder includes a confidence monitor/decoder
• Writes ac3, ec3 & WAV files
• Supports Dolby Digital as well as Dolby Digital Plus
• Perfect compliment to SurCode for Dolby E encoder
• Cross–platform AAX & AudioSuite, for Mac OS & Windows, all on iLok
IN-DEPTH
Why Plus?
Dolby Digital Plus was designed from the ground up to be backward compatible yet broadly adaptable as
your content evolves. It’s scalable enough to address a wide range of delivery methods, from streaming
and download, broadcast and BD to gaming, with outstanding fidelity while still delivering the reliable
performance we’ve all come to expect from AC-3.
For theatrical release and many home theater enthusiasts, 5.1 is not enough. Dolby Digital Plus adds up
to eight main channels to support increasingly popular 7.1 playback. Those extra channels result in more
precise placement and localization for your audience. Thanks to significant gains in coding efficiency, all
this extra capacity is carried as fully discreet, individual channels with no matrixing, so imaging is solid
and repeatable across a variety of speaker configurations and playback conditions. That extra efficiency
also translates into either lower data rates with quality equivalent to Dolby Digital, or superior fidelity at
the same rate or file size as your legacy content.
Being a Dolby–certified product, audio data encoded by the SurCode for Dolby Digital Plus is fully
compatible with all Dolby Digital and Dolby Digital Plus decoders, either in software versions, or as
hardware in CE electronics or stand-alone pro decoders. Your Dolby Digital Plus output can be relied on
to work the way you expect in the consumer’s home, with no guesswork or inconsistencies.
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SurCode for Dolby Digital Plus allows you to deliver an enhanced stream everywhere, with no loss in
quality should conversion to AC-3 be needed.
Complete – End to End
SurCode for Dolby Digital Plus allows an end–to–end, encode/decode cycle for direct control of your
mix’s entire life cycle. Choose 5.1, 6.1 or 7.1 encoding that perfectly matches your delivery platform.
SurCode for Dolby Digital Plus also lets you repurpose 5.1 stems, with 6.1 and 7.1 upmixing. In addition
to a reference grade E-AC-3 encoder and legacy AC-3 encoder, SurCode for Dolby Digital Plus also
provides a facility for downmixing to stereo as well for QC of mixes on consumer equipment.
SurCode for Dolby Digital Plus enables engineers to use SurCode technology in Avid Pro Tools and the
Media Composer Family, making it a powerful, time-saving tool to efficiently mix directly in your DAW.
Now the exact same dialnorm and downmix coefficients of a Dolby Digital/ Dolby Digital Plus consumer
decoding chain can be monitored, and metadata corrected if necessary, saving time and ensuring the
highest quality output during playback.
Tools for Today … and Tomorrow
SurCode for Dolby Digital Plus continues the Minnetonka Audio tradition of supplying discriminating
audio professionals with full featured Dolby codecs. SurCode is the world’s most trusted family of
professional codecs for Dolby formats, with pro–level service and support. As with all members of the
SurCode family, SurCode for Dolby Digital Plus is fully Dolby–certified, allowing the use of associated
Dolby trademarks and branding.
For television, radio, corporate, industrial, games, OTT or Mobile/Hybrid; anywhere, audiences demand
compelling high fidelity content. Dolby Digital Plus is the format of choice, and SurCode for Dolby Digital
Plus delivers with complete confidence.
SPECIFICATIONS
System Requirements
SurCode for Dolby Digital Plus is built for use in Avid’s Pro Tools and Media Composer Family, using the
AAX real-time AAX AudioSuite formats. For specific system requirements and product bundles, please
visit our web site at www.minnetonkaaudio.com.
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