Manual

1.26
Operation Manual
© 2016 Prism Media Products Ltd
Revision 1.01Prism Sound Callia
Conversion quality issues
But surely the sound quality of a digital audio setup can't depend on the choice of digital audio
cabling, so long as all the data bits get through? Sadly, and familiarly, though - it can. Because in
many cases the audio data stream is used to pass the sampling clock as well as the audio data
between equipment. If the receiving equipment gets a clock which has been degraded by a low-
bandwidth interface, and if it uses this clock for A/D or D/A conversion, then the sound quality of that
box will be degraded. This effect is known as 'sampling jitter'. Unfortunately the biphase coding
scheme used in AES3 and S/PDIF is very effective at converting low cable bandwidth into clock jitter.
It should be pointed out that this is an entirely avoidable problem, since any box which relies on
deriving a jitter-free clock for A/D or D/A conversion (or for sample-rate conversion) can take steps to
eliminate incoming jitter - but many don't. The Prism Sound CleverClox technology in Callia does
exactly this, as explained in the Clocking and jitter section. This problem isn't really a cabling issue,
but an equipment design issue. However, in most cases we can't change the design of poor-quality
converters, but we can cover up their problems to some extent with good cabling!
Interference issues
A properly designed copper AES3 or S/PDIF interface will not cause audio-frequency ground
continuity between the connected equipments, so hum loops should not occur. However, high-
frequency ground continuity is essential if EMC legislation is to be met. This means that high-
frequency interference such as from poor-quality switch-mode power supplies (see the Analogue
interconnections section) can equally well be passed through copper digital audio interconnections. If
this is a problem in your system, consider using a TOSLINK connection instead.
Maximising cable performance
In general, the best copper cable for digital audio is the cable with the lowest capacitance, since that
will cause the least loss of bandwidth. For that reason, prefer cables specifically designed for digital
audio, or for analogue video; don't use analogue audio cables - they don't have the bandwidth for
digital audio use, especially at high sample rates. Prefer also the shortest cable, since (all other
things being equal) loss of bandwidth is proportional to length.
Maximising cable bandwidth is important in optimising AES3 and S/PDIF data integrity at high sample
rates such as 192kHz, and in optimising conversion quality in systems which include poor-quality
converters. It is of little importance in protecting the data integrity of low sample rate systems, unless
cable lengths are very long.
We are taught to choose cables of the correct impedance for the job. Whilst this doesn't have a direct
impact on bandwidth, it can have a significant effect on data integrity at high sample rates and where
cabling is short (and let's face it: at 192kHz cables had better be short...) because the reflections
resulting from an impedance mismatch can affect the eye pattern at the receiver horribly. This can be
much worse where non-matched connectors (such as XLRs) are present part way along the cable,
such as in the case of 'breakout' cabled systems. For this reason, it may be better at high sample
rates to use a continuous cable suitably terminated at each end rather than a 'breakout' arrangement.
In summary
· Use good-quality high-bandwidth cables - this means cables specifically designed for digital audio,
or perhaps for analogue video - analogue audio cables are not suitable;
· Don't use cables that are longer than you need;
· At high AES3 or S/PDIF sample rates, consider eliminating 'breakout' connectors in the line by
using a single length of high-bandwidth cable suitably terminated at each end;
· Consider using TOSLINK interconnections in systems where switch-mode interference is a
problem, but remember that poor-quality TOSLINK cables can have very low bandwidth.