Impedance Setting Quick Guide; Wordclock; Working With Stereo Signals - Focusrite TwinTrak Pro User Manual

Ribbon microphones
The impedance of a ribbon microphone is worthy of special mention, as this type of
microphone is affected enormously by pre-amp impedance. The ribbon impedance
within this type of microphone is incredibly low, around 0.2 Ω, and requires an output
transformer to convert the extremely low voltage it can generate into a signal capable
of being amplified by a pre-amp. The ribbon microphone output transformer requires
a ratio of around 1:30 (primary: secondary) to increase the ribbon voltage to a useful
level, and this transformer ratio also has the effect of increasing the output impedance
of the mic to around 200 Ω at 1 kHz.
This transformer impedance, however, is very dependent upon frequency - it can
almost double at some frequencies (known as the resonance point) and tends to roll off
to very small values at low and high frequencies. Therefore, as with dynamic and
condenser microphones, the mic pre-amp input impedance has a massive effect on the
signal levels and frequency response of the ribbon microphone output transformer, and
thus the 'sound quality' of the microphone. It is recommended that a mic pre-amp
connected to a ribbon microphone should have an input impedance of at least 5 times
the nominal microphone impedance.

IMPEDANCE SETTING QUICK GUIDE

In general the following selections will yield the following results:
High mic pre-amp impedance settings
•
Will generate more overall level
•
Will tend to make the low- and mid-frequency response of the microphone flatter
•
Will improve the high-frequency response of the microphone.
Low pre-amp impedance settings
•
Will reduce the microphone output level
•
Will tend to emphasise the low- and mid-frequency presence peaks and resonant
points of the microphone.

WORDCLOCK

Whenever multiple digital audio devices are connected together digitally, all the
devices must be wordclock synchronised to avoid data transfer problems. All devices
must send and receive their data at the same sample rate (e.g. 44.1 kHz) but they must
also have their internal clocks running in sync. This ensures that all units send, receive
and process their data streams simultaneously. Failure to achieve this will mean a drastic
reduction in audio quality, and other unwanted audible artefacts, such as pops and
clicks, may occur. At a sample rate of 44.1 kHz for example, there are 44,100 spaces
every second that need to have samples inserted. If there is a slight drift in one of the
clocks, some of those samples will be 'missed'/will move forward one place, which
results in distortion.
To avoid such problems, every digital system needs to employ wordclock. One unit
should be designated the 'wordclock master', and all others should be designated
'wordclock slaves'. Setting this up is often simple, since most digital transfer formats
include embedded wordclock data (e.g. S/PDIF, AES/EBU, ADAT). Where this is
not the case (e.g. TDIF), wordclock can be provided via a separate wordclock
connection. Note that timecode synchronisation (e.g. SMPTE) is different to
wordclock synchronisation, but equally important. Timecode enables recording and
playback devices to run in sync with one another, and carries a regular series of
absolute time values (hrs:mins:secs:frames). The two timing systems are quite
independent.

WORKING WITH STEREO SIGNALS

There are two basic techniques which can be used to record a sound source in stereo
using two microphones. The first is to use a coincident pair of microphones. This
technique uses a pair of identical directional microphones mounted as close to each
other as possible (typically one above the other) at an angle of up to 90 degrees from
one another, with each microphone feeding one channel. The microphones capture
level differences between the left and right sides of the sound stage – because
directional mics are used, the level varies in direct relation to the physical angle
between the microphones and the sound sources.
The second technique is to use two identical omnidirectional microphones spaced a
fixed distance apart. These microphones capture sounds from different positions in the
sound stage at slightly differing times because of the physical distance between the two
mics, and so they record what is known as 'time-of-arrival' information in the two
channels.
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