Fusion is the 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 is powered by the Axia PowerStation® or StudioEngine DSP mixing engines, which means it is now 100% AES67-compliant. It 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.
- 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.
- 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.
- 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.
- 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
- Any input to any output: +0.5 / -0.5 dB, 20 Hz to 20 kHz
- 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
- 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.
- Threshold: -30dB to 0dB Ratio: 1:1 to 16:1
- Post-processor Trim Level: Adjustable from -20dB to +20dB
- Threshold: -50dB to 0dB Ratio: -30dB to 0dB
- 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
- -10 degrees C to +40 degrees C, <90% humidity, no condensation