Operation Manual

10. Application Guide
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10.5. Digital Audio Connections
Whenever possible, it is preferable to connect a digital rather that analog input signal to the device. This is
particularly relevant if the source signal is already in the digital domain, such as the source from a digital mixing
console or digital distribution system. The primary cause of signal distortion and signal delay (latency) is the digital-
to-analog and analog-to-digital conversion process. Therefore, using digital inputs normally provides higher quality
audio with lower latency.
Two types of digital audio inputs are available: Dante networked multi-channel digital audio, and 2-channel
d interconnections are explained in a separate
The information in this section is supplied for users unfamilia
that the device conforms to established conventions.
10.5.1. AES3 Digital Audio
The original AES/EBU digital audio interface standard was developed by the Audio Engineering Society in
conjunction with the European Broadcast Union. Originally published in 1985, it was revised in 1992 and 2003,
and in its current iteration it is properly designated the AES3 standard.
AES3 is a serial transmission format for linearly represented (uncompressed) digital audio data. It describes a
method for carrying two channels of periodically sampled and uniformly quantized audio signals on a single
twisted-pair cable.
The data format allows for auxiliary data which can be used for information on signal characteristics as well as the
110-ohm twisted pair cabling terminated by an XLR connector. Please refer to section 8.3 for wiring details.
AES3 provides for multiple sampling rates and resolutions of up to 24 bits; this device accepts sample rates from
44.1 to 192 kHz.
10.5.2. System Latency and Delay Compensation
All types of digital audio processing inherently involves a small processing delay referred to as latency. If the
processing chain does not involve analog-to-digital or digital-to-analog conversion, the amount of latency is usually
very small and often may be disregarded.
However, in complex systems involving multiple digital audio components and connections, enough delay may
be generated to cause audio phasing problems. Therefore, the lowest latency is always preferred, and it is always
important to consider system latency delays when calculating and adjusting overall delay for time-aligning multiple
loudspeaker systems.
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10.5.3. Connections and Cabling
10.5.3.1. AES Input Connectors
Two AES-3 input signals (each carrying two audio channels) are connected to the XLR3F connectors labelled
INPUT 1-2 and INPUT 3-4 in the AES/EBU (AES-3) input section on the rear panel. Note that the Input connector
types are identical for the analog and digital inputs, so care must be taken when connecting audio, particularly
when analog inputs are used as a backup signal source. Connectors should be clearly labeled to prevent any
confusion.
NOTE: Never connect a digital signal source to an analog input or an analog signal source to a digital
input.
10.5.3.2. Interconnection of Multiple Units
The AES implementation in PLM+ is designed to be able to daisy-chain AES signals using passive Y-Split cables.
The PLM+ device at the end of a distribution line should be set to TERMINATED; all other PLM+ devices should be
with one DA output per processor, then all terminations should be on. However, if the AES3 is daisy-chained, only
terminate the last processor in the chain.
Please refer to section 10.5.4 for further information.
10.5.3.3. Cable Types and Distance Limitations
All digital connections should be made with 100 ohm balanced cables wired according to the AES3 standard (see
section 8.2.2). Although standard analog microphone cabling may function in limited circumstances, the potential
for problems is greatly increased. AES3 contains a high-speed data stream, and requires an effective bandwidth of
up to 12 MHz, far beyond the 20 kHz required for analog audio.
The distance allowed between a signal source and the PLM+ is dependent on both cable quality and the sampling
rate used. At a 96 kHz sampling rate, any good quality AES3 cable should allow a cable run of 100 meters with no
data losses beyond the capability of internal error correction. The best cables may allow longer cable runs, though
meter length at 96 kHz might extend to 200 meters at 48 kHz, but be cut to 50 meters at 192 kHz.
10.5.3.4. Signal Degradation and Loss
A weak or degraded AES3 signal will exhibit no audible loss of quality as long as the robustness of the data stream
remains above the threshold required for internal error correction. As degradation approaches the threshold,
audible artifacts may be heard, including pops, clicks and momentary dropouts. Any such indications require
immediate attention, as often the window of acceptable data loss between artifacts and complete audio loss can
be very narrow.
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