Orban OPTIMOD-FM 8600, Operating Manual

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OPTIMOD-FM DIGITAL INTRODUCTION
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system the higher the data rate, the less the peak levels will be corrupted by added
noise, so use the highest data rate practical in your system.
It is practical (though not ideal) to use lossy data reduction to pass unprocessed au-
dio to the 8600’s input. The data rate should be at least of “contribution quality”—
the higher, the better. If any part of the studio chain is analog, we recommend using
at least 20-bit A/D conversion before encoding.
Because the 8600 uses multiband limiting, it can dynamically change the frequency
response of the channel. This can violate the psychoacoustic masking assumptions
made in designing the lossy data reduction algorithm. Therefore, you need to leave
“headroom” in the algorithm so that the 8600’s multiband processing will not un-
mask quantization noise. This is also true of any lossy data reduction applied in the
studio (such as hard disk digital delivery systems).
For MPEG Layer 2 encoding, we recommend 384 kB / second or higher.
Some links may use straightforward PCM (pulse-code modulation) without lossy data
reduction. If you connect to these through an AES3 digital interface, these can be
very transparent provided they do not truncate the digital words produced by the
devices driving their inputs. Because the 8600’s output is tightly band-limited to 16.5
kHz, it can be passed without significant overshoot by equally well by any link with
44.1 kHz or higher sample frequency.
Currently available sample rate converters use phase-linear filters (which have con-
stant group delay at all frequencies). If they do not remove spectral energy from the
original signal, the sample rate conversion, whether upward or downward, will not
add overshoot to the signal. This is not true of systems that are not strictly band-
limited to 15 kHz, where downward sample rate conversion will remove spectral en-
ergy and will therefore introduce overshoot.
If the link does not have an AES3 input, you must drive its analog input from the
8600’s analog output. This is less desirable because the link’s analog input circuitry
may not meet all requirements for passing processed audio without overshoot.
NICAM is a sort of hybrid between PCM and lossy data reduction systems. It uses a
block-companded floating-point representation of the signal with J.17 preemphasis.
Older technology converters (including some older NICAM encoders) may exhibit
quantization distortion unless they have been correctly dithered. Additionally, they
can exhibit rapid changes in group delay around cut-off because their analog filters
are ordinarily not group-delay equalized. The installing engineer should be aware of
all of these potential problems when designing a transmission system.
Any problems can be minimized by always driving a digital STL with the 8600’s AES3
digital output, which will provide the most accurate interface to the STL. The digital
input and output accommodate sample rates of 32 kHz, 44.1 kHz, 48 kHz, 88.2 kHz,
and 96 kHz.