and not EBU. AES/EBU is a timecode protocole that has nothing to do with digital audio! Sorry.
AES and S/P-DIF describe two similar protocols for
communicating two-channel digital audio information over a serial
link. They are slightly different in details, their basic format
is almost identical, but there are enough differences that the
two are, for all intents and purposes electrically incompatible.
Both of these digital protocols are described fully in an
international standard, IEC 958, available from the International
Electrotechnical Commission.
AES (which stands for the joint Audio Engineering
Society) is the so-called
"professional" protocol. It uses standard 3-pin XLR connectors
and 110-ohm balanced differential cables for connection (no,
standard microphone cables, not even good quality cables, won't
work, even though it seems they might) and a 5 volt, differential
signal. This is now properly called AES-3, and there is a version
called AES-3id which allows the same format over coaxial cable.
S/P-DIF (which stands for Sony/Philips Digital InterFace, a now
obsolete standard superseded by IEC 958) is the so-called
"consumer" format. It uses what appears to be standard RCA
connectors and cables, but, in fact, require 75-ohm connectors
and cables. Good quality video "patch" cables have proven
adequate (no, standard "audio" patch cords, even excellent
quality versions, have been shown not to work). The signals are
0.5 volts unbalanced.
The actual data streams are very similar. Each sample period a
"frame" is transmitted. Each frame consists of two "subframes",
one each for left and right channels, each subframe is 32 bits
wide. In that subframe, 4 bits are used for synchronization, then
up to 24 bits are usable for audio (the "consumer mode" format is
limited to 16 bits). The remaining four bits are used for parity
(the first level of error detection), validity, user status and
channel status. 192 subframes are collected, and the 192 user
bits and 192 channel status bits are collected into separate 24 8
bit status bytes for each channel.
The channel status bytes are interesting, because they contain
the important control information and the major differences
between the two protocol formats. One bit tells whether the data
stream is professional or consumer format. There are bits that
specify (optionally) the sample rate, de-emphasis standards,
channel usage, and other information. The consumer format has
several bits allocated to copy protection and control: the SCMS
bits.
I am known of using the right digital format cables in each application AND using AES 110 OHM cables for analog audio as well (but not the other way around - using analog mic cable for digital audio/ This is a Major NO).
Now, the notion that all of this is encoded in a standard may be
reassuring, but a standard is nothing but a voluntary statement
of common industry practice. There is a lot of incompatibility
between equipment out there caused directly by subtle differences
between interpretations and implementations. The result is that
some equipment simply refuses to talk to each other. Even THAT
possibility is stated in the standard!

This is a technical post, so tender souls may skip 2 the EASY SECTION @ the bottom.

A few points:

AES / EBU is perfectly ok 2 use when referring 2 digital audio cuz it’s a global standard of the Audio Engineering Society and the Europian Broadcasting Union .

References can b found in both AES and EBU sites.

I think I’ve wrote it b4 ~ a zillion times but what the !@#%:
Because both SPDIF and AES carry clock + audio they r both susceptible 2 jitter.
Best solution is syncing by means of wordclock blah blah blah.

Jitter will b heard only @ the D/A stage, so that’s the best source 4 house clock.

As 4 the cable issue, I find toslink’s (plastic) sound worse than glass’ and coaxial’s (when no wordclock is used of course).
Different media DO NOT SOUND THE SAME in this case. Optics sound “thinner”, due 2 jitter.

SPDIF (IEC 60958) is a jungle. Standard voltage is 0.5 V p>p, which is relatively low, so some computer CD-ROM players use much higher voltage 4 s/n reasons and mark it as “SPDIF”, which will drive any true SPDIF input crazy.

I agree 0.5V is low 4 practical reasons but Sony and Phillips never planned it 4 computer boards or long cables.

1V IMHO would make a good compromise, raising signal quality, and well accepted by standard equipment. Raise ur voltage kids, it will sound better.

EASY SECTION:
(The simple conclusions 4 the easy riders):

Wordclock – good. No wordclock – worse.

Sync wordclock from the D/A internal clock.

If no wordclock used use coaxial rather than optical.

Keep all AES/EBU connections @ 110Ohms, SPDIF @ 75.

Raise (or lower...) SPDIF voltage 2 1V or have some1 do it 4 U.

Don’t drink water after eating fruit.

Peace,

Zooot.