AEQ Phoenix Mercury IP Audio Codec Full Duplex

AEQ Phoenix Mercury

The AEQ MERCURY is a full duplex, Stereo, IP audiocodec allowing for connections in Stereo, Dual or Mono. The codec is exclusively controlled through a simple PC interface, allowing for the remote or local management of one or a series of audiocodecs.

Its very compact design (22 cm (W) x 4,5 cm (H) x 20 cm (D) – two units can be allocated in a 1RU 19” tray) holds a professional audiocodec that is compatible with the majority of audiocodecs from other manufacturers since the unit was been developed in accordance with the international recommendation N/ACIP EBU Tech3326. The AEQ MERCURY supports SIP signalling and the most common encoding algorithms, including the proprietary AEQ LD and optionally AAC encoding modes.

The unit has balanced analogue Audio Inputs and Outputs through XLR connectors as well as digital audio AES/EBU I/O’s.

AEQ MERCURY are delivered readily registered in order to freely use the services of the AEQ SIP Server, simplifying the establishment of connections. The AEQ MERCURY and its bigger companion AEQ VENUS are perfect for STL applications, Radio broadcast networks, outside broadcast contributions or as digital hybrids for IP Telephony.

PHOENIX MERCURY audiocodecs in combination with the ControlPhoenix MANAGEMENT SOFTWARE will put you in the position that you always wanted - IN FULL CONTROL

The ControlPhoenix software allows the control and management of Phoenix Studio, Phoenix Mercury and Phoenix Venus audiocodecs through IP. The equipment can either be part of the same local area network as the PC or workstation that controls them or belong to remote networks. Local equipment are automatically discovered when the application is launched. In order to control the remote units, first of all, they need to be reachable and then the user needs to identify them by their IP address.

IP AudioCodec, stereo, bidirectional: Allows for stereo dual or mono connections. PC User interface provides local or remote control for one or an array of codecs.

Compatible with the most audiocodecs in the market: Supports SIP and the most popular encoding algorithms and is 100% compatible with N/ACIP EBU Tech3326 technical recommendations.

IP Advantages: Adaptive buffer to absorb network jitter. Automatic IP parameter configuration through DHCP. FEC.

AEQ SIP Server: AEQ SIP Server - To simplify IP connections, AEQ offers its own SIP server - and at no cost to you.

Continuous data channel: Allows you to transport embedded data within the audio stream, as an example, for equipment remote control.
Includes encoding algorithms for any purpose. Includes the compulsory algorithms according to N/ACIP EBU Tech3326 technical recommendation plus the AEQ LD low delay algorithms. AAC algorithms can be supplied optionally.

Professional Audio: The unit has balanced analogue inputs and outputs on XLR connectors. Optionally, the unit can be equipped with AES/EBU Digital I/O's.

Reduced foot-print: The unit measures only 22 cm (W) by 4,5 cm (H) and 20 cm (D). Two Phoenix Mercury can be allocated together in 1RU 19"" rack mount.

STL Link (Studio-Transmitter Link): Through IP connections on private VLAN, IP Radio-links, WiMAx, WiFi, ADSL, Cable MODEM, etc., can be used to send the program signal from the production centre to the radio transmission sites.

Radio station networks: The unit can be used to interconnect the audio of the different radio stations in the network through IP. Since the unit is bidirectional the audio can be distributed to the different radio stations in the network and at the same time audio contribution can be established in the opposite direction.

Outside broadcasts and contributions: The unit is able to establish connections with other codecs and smartphone codecs, Phoenix (Studio, Mobile, Mercury, PC, Pocket o Lite) or from other manufacturers that are N/ACIP compliant.

Digital Hybrid for IP telephony: Can communicate through SIP with an IP switchboard (Asterisk or similar) that accepts calls in G711 and HD voice calls in G722 from IP telephony smartphone codecs, IP trunking and through gateways, calls from ISDN and POTS (PSTN). This type of communications is converted into 4W audio and made readily available for On-Air broadcasting.

Analog audio inputs

• 2 x XLR female. 9Kohm. Electronically balanced. Professional line level.
• Analog audio outputs
• 2 x XLR male. Output impedance < 100 ohm.
• Electronically balanced. Professional line level.
• Digital audio I/O
• DB15 connector. AES/EBU interface. SRC on input
• Audio specifications
• Nominal input level. 0dBu.
• Maximum input level. +20dBu.
• Nominal output level. 0dBu.
• Maximum output level. 20dB above nominal level.
• Max. distortion, linear audio <0.03%
• THD + N in SRC @1KHz: -117 dB.
• Dynamic range for linear audio.>105dB.
• Crosstalk <-70dB.
• Frequency response (+/- 0.2dB).
• 50Hz – 15KHz in MPEG 1 L II.
• 20Hz – 20KHz in MPEG 4 AAC*, and for linear audio PCM.
• 50Hz – 7KHz in G722.
• 50Hz - 3KHz in G711.
• Analog I/O: A/D and D/A 24 bit Sigma-Delta converters, 48 kHz max.
• Modes: Mono, Dual Mono, Stereo.
• IP communications interface: Ethernet port. LAN 10/100 base T
• Connector RJ45.
• Fully compliant N/ACIP EBU Tech 3326.
• SIP: Compliant with EBU-Tech 3326 recommendation.
• AUX DATA: DB 9. 9,6 Kbps to 38,4 Kbps flow embedded  in the RTP
• stream.
• USB Type A. Master or Slave operational mode.

Encoding algorithms.

• OPUS with Fs= 48KHz, mono, stereo, with 3 mono and 4 stereo presets.
• Bit rates between 12 and 256Kbps. Audio bandwidth between 6 and
• 20 kHz.
• G711 A law, µ law (64kbps, low delay, 3.5 KHz audio bandwidth)
• G722 (64 Kbps, low delay, 7 KHz audio bandwidth)
• AEQ-LD Fs=16, 32 or 48KHz, mono or stereo. Bit-rates between 64
• and 384Kbps, audio bandwidths between 7 and 20KHz.
• MPEG1 y 2-LII, Fs between 16 and 48 KHz, mono, stereo, dual
• channel and joint stereo. Bit-rates between 64 and 384 Kbps. Audio
• bandwidths between 10,5 and 16,5 KHz.
• AAC-LC* high quality, with Fs=24, 32 and 48 KHz, mono, stereo,
• MsStereo, bit-rates between 32 and 256Kbps., audio bandwidths
• between 9 and 20 KHz.
• AAC-LD* high quality and low delay, Fs=48 KHz, mono, stereo and
• MsStereo. Bit-rates between 32 and 256Kbps, audio bandwidths
• between 8 and 20KHz.
• PCM (linear) very low delay and transparent quality. Fs=48 KHz or
• 32 KHz @ 12, 16, 20 or 24 bits/sample, mono or stereo (bit-rates
• between 576 and 2304 Kbps), audio bandwidths between 15 and
• 20 KHz.
• Smart RTP call-initiation protocol that simplifies connection to compatible codecs.

General Features

• Range of operational temperature: -10 to + 45 º C (14 to 114 º F )
• Dimensions: 1/2 Rack - 1RU, 211 x 44 x 200 mm (8.30" x 1.75" x 7.88")
• Weight: 1 kg (2,2 lbs)
• Power supply 12 V DC. External power supply included Input Power: 110-240V AC, 50-60Hz. 8.5 w.  Auto-range.
• Ventilation. Natural convection – totally silent.
• Apt for in-Studio use.

*  Specifications subject to change without prior notice.

* The AAC encoding algorithms are optional for the Phoenix  AudioCodecs.