Manual
56
XIX.ABOUTTHEALGORITHMS
NX oers a very wide range of encoding algorithms. To some this may seem daunng. Here’s a short guide on how to
choose what’s best for your applicaon:
1 Do I have lots and lots of bandwidth? If you’re running on an enrely unconstrained network like
a campus LAN or local Wi-Fi, Mono or Stereo Linear PCM Mode will oer the highest audio quality
with lowest delay. If you’re hing the public Internet at any point in the link, however, avoid Linear
PCM Mode.
2 Do I require interacvity? If you need to chat back and forth across the link, choose one of our
low-delay algorithms like AAC-ELD or Opus. The deciding factor between these algorithms is digital
bandwidth.
3 Is audio quality the paramount concern? AAC or HE-AAC are the best choices for applicaons that
need excellent audio quality. If delay is also a concern, consider AAC-ELD, which along with Opus,
should be the default choice for radio remote broadcasts. If you are running on an unconstrained
network, Linear PCM or FLAC would be a good choice.
4 Do I need to deliver two unrelated audio signals to the same locaon? AAC, HE-AAC and AAC-LD
oer Dual Mono opons that allow uncorrelated signals (such as dual language broadcasts) to be
combined to a single outgoing stream. Note: It isn’t possible to send one stream to locaon A and
one to locaon B. However, it is possible to send the combined stream to locaons A and B and
have them tap only their respecve channels (although this can be a confusing soluon subject to
operator error).
OPUS
Opus is an audio coding format that is gaining in popularity on the web. It has a good balance between audio quality and
delay over a range of bitrates. It allows interoperaon with web services like WebRTC and apps like Linphone. It’s a good
choice for remote broadcasts for most users.
LINEARPCM
This encoder does not compress audio at all. It uses a 48 kHz sampling rate and simply applies small frames of linear
audio to IP packets. This mode is only useful on high bandwidth LAN or managed WAN environments. Mono Mode
requires a network capacity of 768 kb/s, and Stereo Mode requires a network bandwidth over 1.5 Mb/s.
FLAC
This encoder compresses the audio data using a lossless algorithm. This means that the audio extracted from the
decoder is idencal to the audio input to the encoder, with no coding arfacts. FLAC typically removes 30-40% of the
network data compared to Linear PCM, but the actual data rate is variable and is based on the complexity of the coded
audio. Using FLAC over Linear PCM typically results in a slightly higher (5 ms) overall delay.