ANNEXE A: General characteristics of encoding modes.
OPUS:
OPUS is a completely open and very versatile coding algorithm. Its performance is unrivaled for
voice and audio transmission. It was standarized by Internet Engine Engineering Task Force
(IETF) as RFC 6716, and combines Skype’s SILK codec technology with Xiph.Org's CELT.
This algorithm allows for an excellent audio quality with high compression rate and very low
delay. Phoenix family audiocodecs feature 7 selected OPUS modes covering nearly every
transmission need, from voice to high quality stereo music, with bitrates between 12 and 192
Kbps and audio bandwidth between 6 and 20 kHz. The receiver can automatically adapt to the
particular OPUS mode selected in the transmitting end.
G.711:
ITU encoding standard for processing audio signals in the human voice frequency band,
through the compression of digital audio samples obtained at 8KHz, and typically used in
telephone systems.
Bandwidth: 3.5 KHz
For further information on this subject, consult:
http://www.itu.int/rec/T-REC-G.711/e
G.722:
ITU encoding standard, based on ADPCM algorithms, for processing audio signals in the
human voice frequency band, through the compression of digital audio samples obtained at
16KHz, for greater audio quality and clarity.
This is the internationally accepted mode for two-way communication because of its low delay,
which is why it is the most used standard in commentator and sports broadcasting applications.
Bandwidth: 7 KHz.
For further information on this subject, consult:
http://www.itu.int/rec/T-REC-G.722/e
MPEG LAYER II:
Well known, widely accepted encoding mode that is used when the delay is not important, since
MPEG modes always have a greater delay than G.722 modes. There are 64kbps encoding
modes with sampling rates of 48, 32 or 24KHz, and 128kbps encoding modes with sampling
rates of 32 and 48KHz.
Bandwidth: 10KHz to 15KHz.
For further information on this subject, consult: ISO/IEC 11172-3 and ISO/IEC 13818-3.
AEQ LD+:
AEQ proprietary mode (based on the previous AEQ-LD Extend mode) that combines the low
delay offered by G.722 with the greater bandwidth of the MPEG modes, optimizing these two
aspects.
AAC-LD:
The MPEG-4 Low Delay Audio Codec was designed to combine the benefits of perceptual
audio encoding with the low delay needed in two-way communications. It is a variant of the
MPEG-2 Advance Audio Coding (AAC) format.
For further information on this subject, consult: ISO/IEC 13818-7:1997 and ISO/IEC 13818-3.
PCM (12/16/20/24 bits):
Linear audio without any compress process.
For further information on this subject, consult
http://www.digitalpreservation.gov/formats/fdd/fdd000016.shtml
IMPORTANT NOTE: AAC-LC and AAC-LD modes are optional, they are not available in the
basic firmware version the unit is supplied with. For further information, please consult the AEQ
Sales Department or authorized distributors. Operation with two AAC simultaneously not
supported.
Other different encoding modes can be taken into account according to specific needs of each
client.
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