PrismSound Lyra - Digital Interconnections

1.51
Operation Manual
© 2013 Prism Media Products Ltd
Revision 1.00Prism Sound Lyra
· Use balanced connections where possible; if you must use unbalanced connections, keep them
short;
· Ensure that signals passing between equipment do so at as high a level as is practical;
· If switching interference is heard, try to identify the source equipment by unplugging things one by
one. When you find the culprit, either re-plug it a long way from the audio equipment, or use a
power filter, or both.
7.5 Digital interconnections
It is understandable that little attention is usually paid to the quality of digital audio cabling. We are
used to interconnecting our computer equipment with low-cost cables without mishap, and with
digital audio it's rather logical to assume that no sound quality issues exist since we are simply
moving digital data around.
But the choice of digital audio cabling can be important, because the problems of transmitting digital
audio data aren't really the same as for computer data at all.
Data integrity issues
In general, digital audio interfacing problems are usually (but not always) the result of inadequate
interface bandwidth, which is most often due to the choice of cabling. In extreme cases this can
result in loss of data (and resulting dropouts in the audio) because (unlike computer interconnection
protocols) simple digital audio interfaces such as AES3, S/PDIF and ADAT transmit the data only
once, and without the possibility of error correction. Although there is a possibility that an error can
be detected, this is of little use since no correction or retransmission is possible. So, unlike a
computer interconnection, a mission-critical digital audio connection must ensure that no bit errors
can EVER occur in the data stream EVER! This can be hard to guarantee in the real world,
especially when the system sample rate is high.
This was not really much of a problem when these interfaces were first standardised, since the
bandwidth requirement was quite modest when the maximum sample rate was only 48kHz.
Unfortunately, back then, the use of analogue audio cables for digital audio transmission was actively
encouraged by the choice of XLR and RCA/phono connectors for AES3 and S/PDIF respectively,
even though they typically have poor bandwidth. But for AES3 and S/PDIF, the bandwidth
requirement is directly proportional to the sample rate, since a fixed number of audio and status bits
are transmitted per stereo sample (note that for ADAT/SMUX connections the bandwidth
requirement does NOT rise with sample rate since the number of channels carried is reduced as the
sample rate is increased instead).
Many modern digital audio devices can operate at sample rates as high as 192kHz, and (sad to say)
many digital audio cabling setups don't have the bandwidth to support this reliably. Actually, it's
worse than that - much of the 192kHz-capable equipment has digital audio ports which (either
admittedly or otherwise) don't support reliable operation at 192kHz whatever cable is used. This is
particularly true of TOSLINK ports (the optical variant of S/PDIF).
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
may 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.