Microphone Inputs

• Up to 5 Microphone Inputs with Automix on each Audio I/O Board
• 3 Mic on XLR and 2 Mic on RJ45 (with Talk Box connector/adaptor) all balanced with + 48V Phantom and GPIO on each Audio I/O Board
• Independent ON AIR Lights and cut-off control when opening Control Room and Studio microphones
• PFL button switches to private conversation on Hybrid/Telco/BT mode
• Enable TalkBack feature on any Studio or Control Room microphone
• Parametric Compressors/Expander with enhanced features for MICROPHONE and MONO inputs
• Save and recall microphone presets including Compressor/Expander settings
• Phase inversion on microphone inputs
• MONO inputs assignable as EXT source
• Ducking/AutoFader assignable to all sources (automatic attenuation of music on speech)

 

Oxygen Remoter

Oxygen Remoter is a free software tool included with any Oxygen console. It offers advanced remote control features, allowing users to access setups and information, control faders and buttons, and perform any other operations available on the physical consoles. This tool is particularly beneficial for compact consoles and environments with limited space.

The Virtual Fader technology enables users to assign each input to a virtual fader or channel on the remote interface, which is especially useful for remote channel control. Additionally, transforming the A/B pair into two virtual faders is a smart solution that optimizes the management of available sources.

Oxygen Remoter can control multiple consoles remotely, creating an interconnected ecosystem that is invaluable for remote productions. For instance, some channels can be controlled from a studio while others are managed from the physical console in the control room.

Oxygen Remoter is a powerful tool that enhances flexibility and control for all Oxygen consoles. It can be downloaded from the administration web page of the Oxygen console.

Automix

The Automix feature simplifies mixing activities if many guests are in the studio.

The Automix adds a new source to the existing ones (called “Automix”) and it will be the sum of all the enabled microphones.

The Automix source has all the features of a normal microphone source such as:

• Studio/Control Room loudspeaker control
• Studio/Control Room light control
• Can be used as audio in Private mode for phone calls

Using the Automix source, the user can manage in a very simple way all the microphones of the studio with many benefits:

• Managing all the studio’s microphones with one fader only;
• A dedicated AGC section for each microphone input automatically adjusts the correct level;
• Automatic management of attenuation on microphones not in use, simulating a slight closure of the fader;
• Final compressor with automatic leveling of the Automix source;
• Possibility to use the microphones individually.

This feature simplifies the work of the mixer engineer, especially if he/she is also hosting the show.

Available settings in the Automix menu:

• MIC-1    ON/OFF
• MIC-2    ON/OFF
• MIC-3    ON/OFF
• MIC-4    ON/OFF
• MIC-5    ON/OFF
• MIC-6    ON/OFF (with second audio board)
• MIC-7    ON/OFF (with second audio board)
• MIC-8    ON/OFF (with second audio board)
• MIC-9    ON/OFF (with second audio board)
• MIC-10    ON/OFF (with second audio board)
• DANTE-1    ON/OFF (with Dante™ option)
• DANTE-2    ON/OFF (with Dante™ option)
• DANTE-3    ON/OFF (with Dante™ option)
• DANTE-4    ON/OFF (with Dante™ option)
• DANTE-5    ON/OFF (with Dante™ option)
• DANTE-6    ON/OFF (with Dante™ option)
• Idle Mic Attenuation
• Attenuation Attack Speed
• Attenuation Release Speed
• Compression Threshold
• Compression Ratio
• Compression Attack Speed
• Compression Release Speed
• Autogain Speed
• Autogain Max Gain
• Autogain Max Attenuation

Analogue Inputs

• 2 balanced stereo analogue inputs on RJ45, up to 2 additional balanced stereo analogue inputs (one on XLR and one on RJ45) using other input sources, on each Audio I/O Board
• Channel mode selection: STEREO, MONO, L, R, INV-L, INV-R, INV

Digital Input Line

• 1 AES/EBU Input on XLR with Sample Rate Converter (32 – 192 kHz), on each Audio I/O Board

USB Audio I/O

• 2 Built-in stereo USB (Type-B) (2 Stereo Inputs-1 Stereo Output) Audio Interfaces on each Audio I/O Board, allow you to directly connect the PC to the Oxygen 3000 Plus console, with no need for extra audio cards (the PC detects the console as a digital audio card with 2 stereo inputs and 1 stereo output for simultaneous playout and recording)

Call Management: Telephone Hybrid, Telco, VoIP and Talk Show System

Oxygen 3000 Plus handles calls the way your studio needs them: a built-in telephone hybrid, connections for external hybrids, optional VoIP channels, and the Talk Show System for call-heavy shows.

From the occasional listener call to a show running eight lines at once, call handling stays inside the console.

Telephone Hybrid

• Telephone Hybrid: 1 POTS/PSTN telephone line with Line In and Tel Set connection on each audio I/O board

The built-in telephone hybrid allows direct connection of a telephone line POTS/PSTN on RJ-11C.

Telco

• Telco: 1 in/out N-1 to connect external telephone hybrid with control via TLC/GPIO, on each Audio I/O Board
• Up to 6 Telco using other input sources, on each Audio I/O Board
• GPI and GPO management on max 5 Telco I/O, on each Audio I/O Board
• RING and HOOK on F keys for each TELCO I/O
• N-1 on Telco outputs for all BUS (PGM, SUB, AUX-1, and AUX-2)

A Telco N-1 in/out is also available to connect Oxygen 3000 Plus to an external telephone hybrid. Telco-1 N-1 output is always available on RJ45. TLC commands are available on the RJ45 connector with Hook and Ring interface for an external telephone hybrid. 6 inputs/outputs N-1 are also available (with 4 GPI controls).

VoIP 2Ch (optional)

• 2 VoIP channels with a dedicated web interface and 2 additional audio sources
• A single parameter selects audio streaming or VoIP
• 60-second demo mode to try VoIP 2Ch

Talk Show System (optional)

For call-heavy shows, add the Talk Show System. It manages up to eight telephone lines (POTS/telco, Bluetooth or VoIP) through an integrated web interface, and brings them to air through as few as one or two mixer channels. Your desk no longer needs a fader for every caller.

Call handling is streamlined, with multiple operator support, conference modes, and advanced queue control, with no additional hardware required.

Global RING

Know when a call is coming in, wherever you are in the studio. A global RING state output (GPO) brings together all your telcos, phones, and Bluetooth devices, so an incoming call on any of them can light up lamps in the room.

• Ring status of Telephone 1 and 2 and Telco 1 to 6 is mirrored
• RING and HOOK management works together with Mr Light RGB lamps
• Mr Light RGB lamps can be switched on over Wi-Fi from a dedicated list, and a single Mr Light can be identified among many

Analogue Outputs

• For each Audio I/O Board, 4 Balanced Outputs:
  • OUT – 1 & 2 XLR – Assignable
  • OUT – 3 & 4 RJ45 – Assignable

Digital Output

• 1 AES/EBU output (32 – 192 kHz) on XLR on each Audio I/O Board

Headphones and Local Monitor Outputs

• For each Audio I/O Board, 1 Unbalanced output line on Jack 6.3 for Control Room and Studio Headphones
• Adjustable maximum Control Room and Studio speakers volume
• Adjustable maximum Control Room and Studio headphones volume
• Adjustable CUT or Attenuation mode (-40, -30, -20, -10 dB) on Control Room and Studio speakers
• Independent source selection (PGM, SUB, AUX1, AUX2, EXT1, EXT2) for control room and studio speakers, control room and studio headphones with or without PFL
• Independent Talk Back management on Control Room and Studio speakers
• Independent Talk Back management on Control Room, Studio and Guest headphones
• Selectable source for Guest headphones (PGM, SUB, AUX1, AUX2, EXT1, EXT2)
• Same source selection for Control Room and Studio headphones, with independent level

GPI/GPO

• GPI: 9 on each Audio I/O Board, for talkback, Telephone Hybrid, or other features
• GPO: 9 on each Audio I/O Board, to control OnAir lamps or provide HOOK and RING commands for external telephone Hybrid
• User custom GPIs (e.g. Ring, Talkback, channel ON/OFF, etc.)
• User custom GPOs (e.g. Hook/ONAIR, Studio-Light, Control Room-Light)

Surface & Management

• Direct Firmware update via Internet
• WEB UI for Console configuration and service features (Save, restore, update, logs, etc)
• Conditional Access (4-digit PIN) to limit console configurations to authorized users
• Save and recall EQ presets (up to 10)
• New real-time EQ algorithm with graphic
• Fast and accurate Meters on 2.2” TFT IPS displays
• Save and recall control panel configuration with 10 Snapshots
• Save and recall total console configuration. The configuration can be saved to internal memory (microSD), USB, and PC (via WEB interface)
• Gain adjustment knobs with adaptive speed
• Dimmable buttons light
• Adjustable fader ON/OFF threshold (from -50 dB to 0 dB)
• CountUp/Timer on studio and control room microphones on the built-in 7″ display

Displays

On the OXYGEN 3000 Plus, AxelTech reinvented the studio engineer-to-console experience.

The 7″ TFT IPS graphic display shows all the airing parameters and, with the 4 rotary knobs, allows the fast and intuitive programming of all the mixer’s parameters.

Selectable home page layouts are available with many theme icons. The default page shows the levels and the A & B source names for any fader.

Four 2,2 ”TFT IPS displays have been implemented for immediate and complete monitoring of the outputs.

The 2,2 ”TFT IPS displays on top of each fader show the channel status, levels, sources, and main settings.

HDMI Output

The HDMI output connects the console to an HD display monitor, and it can monitor all  the following information:

• Audio input source levels
• 4+1 BUS output levels
• A/B source names assigned to 10 channels
• Display On-Air sources
• Microphone on timer (control room & studio)
• Large Clock
• Audio output level (Loudspeakers & Headphones)
• On-Air microphone
• Telephone active
• HDMI output combined with 4 rotary knobs allows fast, intuitive programming of all mixer parameters.
• The programming GUI has been designed in order to be very easy and intuitive.
• Selectable home page layouts are available, with many theme icons.

HDMI output can be customized with the user’s logo. Selectable display layouts are available.

BT

Oxygen 3000 Plus has a BT module on each Audio I/O Board. This allows airing the calls made via GSM, Skype, Viber, and WhatsApp with a Smartphone, a tablet, or a PC. Oxygen 3000 Plus has two BT connection modes: bidirectional to support phone calls; monodirectional to put on air high-quality stereo audio (A2DP) streamed by BT sources.

With BT calls, PFL enables PRIVATE conversation mode.

Built-in Decoder and Streamer

In addition to analogue, digital and AoIP inputs and outputs, Oxygen 3000 Plus connects to the IP world. The built-in decoder receives an audio stream (stereo or mono) as an input, and the streamer sends an output audio stream (stereo or mono).

The streaming channels can also be switched to VoIP. With the optional VoIP 2Ch, a single parameter lets you choose between audio streaming and VoIP.

Supported CODECs:

Input (CBR & VBR):

• AAC (AAC-LC): 32-256 kbps, 16/32/44.1/48 kHz
• Linear PCM – 16 bit, 16/32/44.1/48 kHz
• MP3 (MPEG1-Layer 3): 32-320 kbps, 16/32/44.1/48 kHz
• MP2 (MPEG1-Layer 2): 32-320 kbps, 32/44.1/48 kHz
• OGG VORBIS: 32-256 kbps, 16/32/48 kHz
• OPUS: 6-320 kbps, 48 kHz

Output (CBR):

• AAC (AAC-LC): 32-256 kbps, 16/32/48 kHz
• Linear PCM – 16 bit, 16/32/48 kHz
• MP3 (MPEG1-Layer 3): 32-320 kbps, 16/32/48 kHz
• MP2 (MPEG1-Layer 2): 32-320 kbps, 16/32/48 kHz
• OGG VORBIS: 32-256 kbps, 16/32/48 kHz o OPUS: 6-320 kbps, 48 kHz

Supported Streaming Servers:

• Icecast: MP3, OPUS and Ogg Vorbis

Network and Transport & protocols:

• Transport: Real-time Transport Protocol (RTP), Real-time Streaming Protocol (RTSP)
• Protocols: IPv4, IPv6, TCP/IP, UDP, RTP, RTSP

Casting modes: Multicast and Unicast

Key applications:

• STL and SSL links, DVB S/T audio, WEB radio
• Audio streaming over the internet

Talk Show System

Transform Your Live Show Production with Seamless Phone Integration

The Talk Show System revolutionizes how radio and broadcast professionals manage incoming calls, delivering unprecedented control and professionalism to your live productions. This powerful software option integrates directly with the Oxygen 3000 Plus console, transforming complex multi-line phone management into an intuitive, streamlined workflow.

Why Talk Show System Changes Everything

Professional Call Management Made Simple. Gone are the days of juggling multiple phone hybrids and cluttered mixing surfaces. The Talk Show System consolidates up to 8 telephone lines into just one or two mixer channels, dramatically simplifying your workflow while enhancing broadcast quality.

Integrated Web Interface: Our dedicated web interface runs seamlessly alongside your Oxygen 3000 Plus, providing three independent operators with customized access rights and specialized functions. Whether you’re screening calls, managing queues, or conducting live interviews, each team member has the tools they need.

Flexible Line Management Support for all telephone line types – POTS, BT, TELCO, and VoIP – means you can utilize your existing infrastructure while expanding your capabilities up to 8 simultaneous lines.

Key Features

Multi-Line Flexibility

• Up to 8 telephone lines supported
• All line types: BT, internal POTS, TELCO (POTS, BT, VoIP)
• Virtual channel routing to TalkShow 1 and TalkShow 2 sources

Advanced Call Modes

• Alternating Queue Mode: Seamlessly cycle through 8 lines
• Conference Mode: Connect up to 8 callers simultaneously with flexible grouping
• Hybrid Mode: Perfect for remote guests with listener call-ins
• AUTO and MANUAL operation modes

Professional Workflow Tools

• Call queue management with comprehensive caller database
• Real-time caller information with mood indicators
• Inter-operator chat system for seamless coordination
• Private mode (PRIV) for PBX-free environments
• Dedicated operator access rights for specialized roles

Streamlined Surface Control

• Reduced fader requirements – potentially downsize to smaller mixing surfaces
• Clean interface – no telephone lines cluttering your mixer layout
• Instant call switching directly from the web interface

Who Benefits from Talk Show System

Radio Stations managing high call volumes for talk shows, contests, and listener interaction

Broadcast Facilities requiring professional phone integration without standalone hybrid costs

Production Houses needing flexible, scalable phone management solutions

Podcasters and Streamers looking to elevate their production quality with professional call handling

Competitive Advantage

Standalone solutions come with high costs, limited mixer integration, and often lack hybrid connectivity options. Many integrate only with high-end consoles, making them inaccessible for mid-market applications.

Talk Show System delivers:

• Superior integration with Oxygen 3000 Plus
• Multi-line hybrid support (BT/POTS/TELCO with VoIP capability)
• Affordable pricing compared to standalone solutions
• No additional hardware required – pure software activation

Technical Specifications

Compatibility: Oxygen 3000 Plus exclusive software option

Line Capacity: Maximum 8 telephone lines (internal or TELCO)

Connection Requirements: Audio and control (HOOK/RING) as per mixer specifications

System Requirements: Medium-performance PC with Full HD monitor, Chrome/Firefox browser

Recommended Configuration: Minimum 2 audio cards for optimal performance

Get Started Today

The Talk Show System transforms your Oxygen 3000 Plus into a professional broadcast communication hub. Contact our team to learn how this powerful option can elevate your production quality while simplifying your workflow.

Ready to revolutionize your call management?

Software activation only – no additional hardware required

VoIP Channels

Add two VoIP channels to your Oxygen 3000 Plus, with a dedicated web interface and two additional audio sources. VoIP calls travel over the same network as the rest of your studio, not over dedicated phone circuits. A single parameter lets you choose between audio streaming and VoIP, and a 60-second demo mode lets you try VoIP 2Ch straight away.

Redundant Power Supply

Oxygen 3000 Plus is equipped with an external Universal Switching Power Supply 12V (with 4-pole XLR connections) able to operate with a voltage between 90Vac and 264Vac and a frequency between 50 and 60Hz. This PSU can adapt itself to the operating voltage of any country in the world. Optionally, the console can be equipped with a redundant power supply: in this configuration, the fault of one of the PSUs is promptly noticed on the 7” display.

Talk Box

Talk Box is very simple to connect to the On Air Console: it takes 2 RJ45 cables of the length that is needed to connect it to the mixer. It is possible to connect up to 4 Talk Boxes on each Audio I/O Board. The Talk Box manages a direct connection of the microphone and its own On-Air light. It is also possible to connect it to the studio’s On-Air Light. The Talk Box has a headphone amplifier with two 6.3mm outputs. Through the two backlit buttons, it is possible to activate and deactivate your own microphone (On/Off/Cough) or enable talkback to the console.

Dante™ AoIP Interface

Oxygen 3000 Plus can be equipped with a Dante™ interface (also supporting AES67 and SMPTE ST 2110-30 transport protocols) which provides 16 Inputs and 16 Outputs or 32 Inputs and 32 Outputs on a dedicated Ethernet port, with independent Level Control and Sample Rate Conversion.

Dante™ AoIP interface is installed at the factory and has to be requested at the time of order.

Cable Adapters

RJ45 adapters are available for:

• Mic with GPIO
• Line In
• Line Out with GPIO
• Telco with GPIO

Analog Balanced Microphone Inputs
Connector	RJ45 & XLR Balanced – EMI Suppressed
Input Impedance	2,4 KΩ
Nominal Input Level (sensitivity)	-9/-66 dBu
Max Input Level (clipping point)	+9 dBu
A/D conversion	24 bit / 48 Khz
Signal To Noise Ratio (referred to peak level)	>110 dB
THD+N	<0,01%
Analog Gain	Adjustable +0 ÷ +60 dB (3dB step)
Phantom Power	+48V

Analog Balanced Stereo Inputs
Connector	XLR & RJ45 Balanced – EMI Suppressed
Input Impedance	10 KΩ
Nominal Input Level (sensitivity)	0 dBu
Max Input Level (clipping point)	+18 dBu
A/D conversion	24 bit / 48 Khz
Frequency response	+/-0,5 dB from 20 Hz to 20 kHz
Signal To Noise Ratio (referred to peak level)	>110 dB
Stereo Separation (referred to peak level)	>90 dB
THD+N	<0,002 %

Analog Balanced Telco Input
Connector	RJ45 Balanced – EMI Suppressed
Input Impedance	10 KΩ
Nominal Input Level (sensitivity)	0 dBu
Max Input Level (clipping point)	+18 dBu
A/D conversion	24 bit / 48 Khz
Signal To Noise Ratio (referred to peak level)	>110 dB
THD+N	<0,002%

Digital Input
Connector	Balanced on 1 XLR – EMI Suppressed
Input Impedance	110Ω
Standard	AES3
Audio Sample Rate	32/44.1/48/96/192 KHz with SRC
Resolution	24 bit
Dynamic Range (Converter Values)	124dB

Analog Balanced Stereo Outputs
Connector	RJ45 Balanced – EMI Suppressed
Output Impedance	23 Ω, nominal 600 Ω
Nominal Output Level	0 dBu
Max Output Level (clipping point)	+18 dBu
D/A conversion	24 bit / 48 Khz
Signal To Noise Ratio (referred to peak level)	>110 dB
Stereo Separation (referred to peak level)	>90 dB
THD+N	<0,002 %

Dante I/O
Sample Rates (16x16 Versions)	44.1/48/88.2/96 kHz
Sample Rates (32x32 Versions)	44.1/48/88.2/96 kHz
Audio Flows In/Out	Up to 32x32 simultaneous streams
Digital Audio Formats	TDM, I2S
Audio Transport Formats	Dante Audio over IP, AES67 RTP, SMPTE ST2110-30 RTP (enrolled devices)
Sample Bit Depth	16, 24 or 32 bits per sample
Audio Buffering	Up to 2000 samples per channel
Clocking	Onboard word clock or external word clock
Form Factor	Card edge module. 4.5cm x 6cm (1.75” x 2.4”)
FPGA	High performance Xilinx Spartan6 FPGA
Microprocessor	Soft-core Microblaze processor
Clock	High quality, low jitter onboard SiLabs
Ethernet	Standard RGMII/MII interface for Ethernet PHY or switch chip
Power	3.3VDC @ 2W max
Physical Connector	Mini-PCI
Control Interfaces	SPI Master and Slave; GPIO; I2C
Network	RGMII/MII

USB Audio Digital I/O
Connector	USB Type B – EMI Suppressed
Playback And Recording Sample Rate	SRC 44.1-48 KHz
Resolution	16 bit
Available Stereo Channels	2 stereo USB - 2 Inputs & 1 Output on each Audio I/O Board

Analog Balanced Talkbox Outputs
Connector	RJ45 balanced – EMI Suppressed
Output Impedance	100ohm, nominal 600ohm
Nominal Output Level	0dBu
Max Output Level (clipping point)	+14dBu
D/A conversion	24 bit / 48 Khz
Signal To Noise Ratio (referred to peak level)	>110dB
Stereo Separation (referred to peak level)	>90dB
THD+N	<0,05%

Analog Balanced Telco Output
Connector	RJ45 Balanced – EMI Suppressed
Output Impedance	23 Ω, nominal 600 Ω
Nominal Output Level	0 dBu
Max Output Level (clipping point)	+18 dBu
D/A conversion	24 bit / 48 Khz
Signal To Noise Ratio (referred to peak level)	>110 dB
THD+N	<0,002 %

Digital Output
Connector	Balanced on 1 XLR – EMI Suppressed
Input Impedance	110 Ω
Standard	AES3
Audio Sample Rate	32/44.1/48/96/192 KHz
Resolution	24 bit
Dynamic Range (Converter Values)	124 dB

PSTN Interface
Connector	RJ11
Transhybrid loss	>20 dB

System
Audio Core	Analog Devices ADAU1452 32bit 294 MHz fixed point DSP
Audio CODECs	Cirrus CS42448 24 bit/192 kHz
LAN Connection	RJ45 - 1 Gbit
Nominal Delay (analog input to analog output)	0,7 ms
GPIO Inputs/Outputs	4 GPI/4 GPO on DB9; 4 GPI/2 GPO on Mic2 & Mic3 RJ45; 2 GPO on Out3 & Out4 RJ45; 1 GPI/1 GPO on Telco RJ45
Communication Port	2xUSB type-A, 2xUSB type-B , 1xLAN, 1xHDMI
Operating Temperature	0°C ÷ 40°C

Dimensions
8 Faders Dimensions (W; H; D)	584 mm; 113,5 mm; 495,5 mm
12 Faders Dimensions (W; H; D)	764 mm; 113,5 mm; 495,5 mm
16 Faders Dimensions (W; H; D)	944 mm; 113,5 mm; 495,5 mm
Weight	15 to 25 Kg

PSU
Type	External Universal Switching Power Supply 12V with 4 poles XLR connections - Optional Redundant
Power Supply	90-264 VAC / 50-60 Hz / 120 W