Sources (Audio Xnode) - Telos Alliance xNode Serie Bedienungsanleitung

The first column is the Name field, used for documenting the physical output or network stream input, which is best
labeled as a destination. For example, SpeakerAMP could document that the destination is to the speaker amplifier.
The Channel column is used to define the network source being called for to this destination. You may type in the
channel number or use the browse button to search the network for the desired source. The Type column defines the
stream type that is being called. Livewire stream types are known as "From Source," "To Source," or in some cases
"Surround." The "To Source" option is a special dynamic return audio known as a backfeed. A backfeed is a channel
automatically generated by Axia consoles for talkback, mix-minus, etc. The Gain column provides adjustment for
the destination from the range of -72dB to +24dB.

Sources (Audio xNode)

The Sources page typically defines the physical audio inputs of the xNode. The corresponding network streams are
given a name and a unique channel number that is used to distribute through the network. The xNode provides an
auto-numbering process when you ID the xNode through the node ID process described in Chapter 2.
Source Name column is used to identify the physical input or network stream generated by the xNode (for documen-
tation sake) and is also used in source advertisement through the network.
Channel column is the unique number for each source. This will auto-populate based on the ID given to the xNode.
It is critical that each source in the network has a unique number, not shared with any other source.
Stream Mode column defines how the audio is packetized. Live stereo is the original Livewire stream intended for
microphone or other live audio content that needs a lower delay. A packet contains 12 samples or 250us of audio.
Low Latency Stereo is an option for support of Ravenna and AES67 streams. The packet contains 1ms of audio (48
samples). Standard Stereo is the high efficiency stream that contains 240 sample (5ms of audio), maximizing the
packet size at 1522 bytes. This option is intended for suitable sources, such as playback machines that can use the
higher efficiency with slightly more latency.
Input Gain column provides adjustment of -72dB to +24dB for line level audio and +77dB for microphone audio
inputs. When a source is enabled through the stream mode, an RTSP link will appear under the Source row. This
link can be given to a media player that supports RTSP as well as 24-bit PCM streaming audio for direct auditioning
of the audio stream. VLC 1.1.11, for instance, is a cross-platform media player that supports RTSP. An AES67 SDP
stream description is also provided.
Unicast Link
Unicast Link page allows configuration of multichannel audio link between locations connected with an Ethernet
network without support for IP multicasting. In particular it can be used as a studio transmitter link. The page func-
tions as a configuration wizard and applying the settings makes changes in audio routing setup and other options
presented on the Sources, Destinations, and QoS pages.
Unicast Link is by default bidirectional. The connection protocol supporting Unicast Link is Real Time Streaming
Protocol (RTSP), which is a TCP/IP protocol. Audio is sent using unicast UDP/IP protocol. A basic setup involves
setting one xNode to Passive, the first radio button. The second xNode will be configured with the IP address of
the remote "Passive" xNode in the "Connect To" field. Connection Status area provides diagnostic information of
the established link. When the connection is successful, the icons turn green and Stream active / RTSP OK status is
displayed. The Setup interface allow you to save six presets, which can be different remote nodes. Selecting a preset
is a fast way of changing the link destination.
© 2017 Telos Alliance - Rev 2.0.1