Orban OPTIMOD-FM 8600Si, Operating Manual

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OPTIMOD-FM DIGITAL INTRODUCTION
1-17
Because the 8600Si 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 8600Si’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.
If you use a lossy codec between the Optimod’s output and the transmitter’s input,
the codec must see a “flat” signal to ensure that its psychoacoustic model works cor-
rectly. Hence, the Optimod output driving the codec must be set to
F
LAT
,
which ap-
plies deemphasis to the output of the Optimod’s preemphasis processing. 50
μ
s or
75
μ
s preemphasis must then be applied to the codec’s output signal. This usually oc-
curs in the FM transmitter’s stereo encoder or in a stand-alone stereo encoder like
an Orban 5518.
Some links may use straightforward PCM (pulse-code modulation) without lossy da-
ta 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 8600Si’s FM output is tightly band-limited
to 17 kHz and its HD output is limited to 20 kHz, these signals can be passed without
additional overshoot by equally well by any link with 44.1 kHz or higher sample fre-
quency.
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
8600Si’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 pre-
emphasis.
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 8600Si’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.