Codec Tieline Gateway, User Manual

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© Tieline Research Pty. Ltd. 2021
Gateway and Gateway 4 Manual v1.2
Overview of aptX Enhanced Audio Coding
aptX® Enhanced audio coding is used by thousands of radio stations to deliver very low delay audio
for IP broadcasts and is ideal for high quality studio-to-transmitter links and audio distribution. It
delivers outstanding audio quality with exceptionally low delay across a range of IP networks.
32kHz or 48kHz sample rates are available at either 16 bit or 24 bits per sample. aptX Enhanced
has a minimum connection bit rate of 128kbps per channel and offers 10Hz to 24kHz frequency
response. 24 bit, 48kHz aptX Enhanced at the maximum bit rate of 576kbps delivers >120dB of
dynamic range.
aptX® Enhanced is supported over ISDN at the following sample and bit rates:
Encoding Bit rate Required B Channels Required
aptX® Enhanced Mono 16 bit, 32 kHz 128 kbps 2
Overview of Opus Algorithm
Opus is a highly versatile open source audio coding algorithm. It incorporates technology from the
well-known SILK and CELT codecs to create a low latency speech and audio codec. It is a variable
bit rate algorithm ideal for live broadcast situations because of its capacity to deliver high quality,
real-time Audio over IP (AoIP) at low bit rates. Visit
http://www.opus-codec.org for more info.
There are three Opus algorithm configurations available:
Algorithm Recommended connection for on-air use
Opus Voice High quality low bit rate remotes (9.6kbps -64kbps)
Opus Mono Very high quality mono remotes, STLs and audio distribution (48kbps -128kbps)
Opus Stereo Very high quality stereo remotes, STLs and audio distribution (64kbps -256kbps)
How do I choose the right algorithm?
The algorithm you select will not only affect the quality of the broadcast but it will also contribute to
the amount of latency or delay introduced. For example, if MP2 algorithms are used, program
delays will be much longer than when using Tieline Music or MusicPLUS algorithms. This is due to
the additional inherent encoding delays involved when using MP2 algorithms. This can be a major
consideration for live applications that integrate remotes into a broadcast. The algorithm you select
to connect with will also depend upon:
·
The codecs to which you are connecting (Tieline versus non-Tieline)
·
Whether you are creating multi-unicast connections.
·
Whether you are connecting using SIP or not.
·
The uplink bandwidth capability of your broadband connection.
Important Notes: Music and MusicPLUS algorithms cannot be used over SIP connections.
Use MP2 algorithms at 64kbps mono or 128kbps stereo for high quality connections when
using SIP, or use G.711 and G.722 if required. Tieline G3 codecs do not support
connections using AAC, aptX Enhanced and Opus algorithms and will default to MPEG
Layer 2 if an incoming connection is configured to use these algorithms.
It can be a good idea to listen to the quality of your program signal using each algorithm and to see
how it sounds when it is sent at different connection bit rates (as well as different FEC and jitter-
buffer millisecond settings). This will assist you to determine which is the best algorithm setting for