AVT MAGIC AC1 Go Compact Stereo IP Audio Codec

The MAGIC AC1 Go is a compact stereo IP audio codec that provides an additional independent command channel via G.711/G.722.The system can be used for AoIP dial-up connections or IP leased lines. In AoIP mode, the system can register with 5 different SIP servers. For outgoing calls, the respective SIP server can be selected, incoming calls can be accepted automatically.

For reliable audio connections, Secure Streaming is integrated to guarantee highly reliable transmission by means of redundant transmission. The streams can be transmitted with delay to avoid failure with of inaudible switching even in case of burst errors.

The system supports the coding algorithms G.711, G.722, ISO/MPEG Layer 2, Opus, FLAC and PCM 16/20/24 bit in the standard version.
Optionally, the audio codecs can be extended with MPEG Layer 3, AAC-LD/AAC-ELD and AAC-LC+V1/V2.

The audio programmes can be input or output via an analogue or digital stereo interface (switchable).
The unit has an Ethernet interface with VLAN support for control and IP audio transmission.
Two RS232 interfaces are available for additional data and for control purposes. 4 TTL-GPIOs and 2 relays can be used for signalling.

The system also supports the Ember+ and DHD SetLogic protocols to exchange control and signalling commands with e.g. mixing consoles via 64 virtual GPIOs. Accessing status information and triggering of functions in the unit is possible via Ember+.
The system can be integrated into a network management system via the integrated SNMP protocol.

MAGIC AC1 Go is designed as a ½ x 19″ system with external power supply.

The audio codec can be controlled via the front keypad with display or with the Windows PC software included in the delivery, with up to 5 workplaces being able to access a system simultaneously.
Multiple systems can also be centrally managed, operated and monitored via the Multi-Control software, which is also included.
The transmit and receive levels of all transmitted or received audio channels are displayed, including general system information such as IP addresses and alarms.

The internal system log file allows tracking of errors even without a connected PC and can be downloaded from the system at any time via the management software and displayed in an easily understandable way. For date and time information, the system supports NTP synchronisation.
Monitoring and alarming can also be done via SNMP. Traps can be reported to up to four network management systems.

In addition to the coding algorithms supported by default, the ISO/MPEG Layer 3 Upgrade, the FhG AAC-LD/ELD, the AAC-LC and the HE-AACV1/V2 Upgrade are optionally available via software upgrades.

Features:

• High quality audio transmission with up to 20 kHz.
• 1 x analogue or 1 x digital stereo Audio input/output (switchable)
• EBU Tech 3326 compliant (AoIP standard) and compatible with all VoIP telephones
• Simultaneous registration on five SIP servers
• SIP or leased line mode
• Secure Streaming
• Independent command channel (G.711/G.722)
• Control via Windows PC software
• Optional: 4 x AES67 channels via software upgrade

• Technical Data Line Interfaces: 1 x LAN
• Coding Algorithms: G.711,G.722,PCM 16/20/24,MPEG Layer 2,Opus,FLAC,
  Optional:MPEG Layer 3,AAC-LD,AAC-ELD,AAC-LC,HE-AAC V1,HE-AAC V2
• Sampling Frequencies: 8, 16, 24, 32, 48-kHz (depends on Codec)
• Data Rates: 16–2304-kbit/s (depends on Codec)
• Audio Inputs: 1 x Stereo analogue XLR 1 x Stereo digital AES SUB-D 15-pin Audio Outputs: 1 x Stereo analogue XLR 1x Stereo digital AES SUB-D 15-pin
• AoIP Channels: 4 x AES67 RX (2 streams) 4 x AES67 TX (1 stream) (optional)
• Secure Streaming for IP Leased Line Mode: Yes
• DHD SetLogic/ Ember+ protocol”Yes
• Control Interface: GPIO (4 x TTL, 2 x Relays) 1 x LAN
• TTL Interface (GPIO): 6 x Input/Output, programmable SUB-D 25-pin
• Relay Contacts: 6 x NC/NO, programmable SUB-D 25-pin
• Power Supply: external 12 V
• Redundancy: 12 V DC external power supply (optional)
• Dimensions: ½ x 19”, 1U