Lab.gruppen PLM 20K44 - Manual (Page 80)

10. Application Guide
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PLM+ SERIES Operation Manual rev 1.0.0
10.5. Speaker Configurations
Connecting two speakers in parallel to a PLM+ power output presents a load to the amplifier which is half the
impedance of that presented by one speaker. Therefore, the current that two speakers will attempt to draw from
the output stage is double that for one speaker, and this higher current may be sufficient to cause the Current Peak
Limiter to become active. The more speakers connected to an output in parallel, the lower the impedance and the
higher the current draw.
Multiple loudspeakers may be driven by a PLM+ power output more satisfactorily if a series-parallel wiring
configuration is adopted. Please ensure care is taken to match polarity correctly.
When using series-parallel wiring, the nominal impedance is the same as with one speaker; however, the principle
of power sharing still applies, and it is not possible to get the amplifier section to deliver more than its rated power.
Note: Nominal loads as low as 2 ohms are supported by the PLM+. However, a 2 ohm nominal load
has impedance dips at its resonances below 2 ohms; in such cases it is likely that the resulting higher
current will cause CPL to activate.
10.6. 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
digital audio via the AES3. Dante-based system configurations and interconnections are explained in a separate
document, the Lake Network Configuration Guide.
The information in this section is supplied for users unfamiliar with AES3. Users already familiar with AES3 will find
that the device conforms to established conventions.
10.6.1. 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.