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AES67 is an open standard for Audio over IP
developed by the Audio Engineering Society
(AES). It sets a minimum standard for audio
streaming and synchronisation in Audio over IP
networks such as Dante, RAVENNA, Livewire+ and
others.
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The optional AES67 functionality of the device is
configured on the AES67 page.
LAN INTERFACE: Select the IP address of the unit
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to be used for AES67.
CHANNELS: Set the number of AES67 mono audio
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channels.
TRANSMISSION: This section defines the AES67
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stream that the unit sends.
SAP STREAM NAME: Enter any text. The stream is
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announced on the network using that name.
RTP UDP PORT: Set the local UDP port for the
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AES67 stream. (Default: 5300)
AES67 (2)
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AUDIO MODE: The audio signal is PCM encoded.
The bit depths can be set here:
L16: 16 bit/sample
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L24:24 bit/sample (Recommended for connections to
Dante devices.)
SAMPLING RATE: Set the sampling frequency of
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the audio signal.
32 kHz
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48 kHz
ADDRESS MODE: AES67 streams are multicast on
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the network. There are two modes for entering
the multicast IP address:
MANUAL: You can freely enter the complete IP
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address.
AUTO: The IP address is derived from the IP address
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of the LAN interface of the device. Only the second
byte of the IP address must be entered manually.
IP ADDRESS: Enter the entire IP address or the
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second byte of the IP address, depending on the
ADDRESS MODE set.