MTM48.

10mm structure

1/30/07

4:16 PM

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TE$ MI$(TE
MASTER No99

Gain structure
Eliminating extraneous noise is a primary task in the studio, but we may be inadvertently
generating it ourselves. Grant Bridgeman explains the importance of good gain structure...

igital recording
systems running at
24-bit resolution are
designed to capture
truly high-quality
recordings, ensuring
the best possible signal-to-noise
ratio and dynamic range. That
said, signal noise and internal
distortion are still common
frustrations, often leading
anxious home recordists to go
on expensive and sometimes
unnecessary spending sprees.
In the majority of cases, though,
its not the equipment thats at
fault; rather, its the gain structure
of our recordings and mixes that
needs to be reviewed.

Structural integrity
Correct gain structure within
either an analogue or digital
system will ensure that the end
result utilises all of the available
dynamic range, has a minimal
noise floor and doesnt
introduce unintentional
distortion to the signal. This
may appear to be a very generic
statement that can cover all
good recording practices, but it
is very specific when referring to
gain structure as any distortion
added to the signal path at one
stage cannot be removed at any
subsequent stage.
Microphone

Preamp

Every stage in the signal path

should be set so that each
processor (be it hardware or
software) receives the correct
input level and therefore operates
optimally within its own dynamic
range. A good example is that of
a reverb effect being used on an
auxiliary buss channel.
For this example, lets
assume that we want to add a
small amount of reverb to the
source signal. This can be
achieved in two ways. Firstly,
we can send a small signal from
the channel strip (via the aux
send) to the reverb unit and set
the outputs of the reverb to
unity gain or higher to bring the
signal to the correct level within
the mix. The problem with this
approach is that the signal level
for the reverb is very low in
terms of the input range of the
device, and consequently close
to the noise floor, giving a poor
signal-to-noise ratio. As the
output of the device is very
quiet, it is likely that its level will
have to be boosted beyond
unity gain, raising the noise
floor of the reverb along with
the signal.
The second approach is to
have the aux send from the
channel strip set at (or near)
unity gain, ensuring that the
EQ

Volume fader

Irrespective of what each device in your signal chain actually does, it is important to
ensure that the output level of each device is suitable for the following device.

38 March 2007 MusicTech MAGAZINE

Original audio signal

The same signal after

digital clipping. The
loss of information
causes distortion.

A clipped signal that

has been boosted. The
unwanted distortion is
boosted proportionally.
Once a signal has clipped, boosting the signal at a later stage will merely result in a
louder clipped signal. Preventing clipping from occurring in the first place is key.

processor receives a sufficiently

high input signal (watch out for
clipping). The mix level of the
effect is controlled by the return
level. This means that the
reverb processor is operating at
a level within its dynamic range
and the noise floor is much
further from the signal.

The weakest link

The concept behind good gain
structure states that each stage
should be working capably
within its operating range, but
unwanted signal distortion can
still occur at any point in the
signal path. Microphone
preamps, for example, are
particularly prone to internal
distortions, especially those
found on lower-end mixing
consoles. It is feasible for the
microphones themselves to
distort (check the SPL levels for
1% THD on the mics spec

sheets), but it is far more likely

to be the mic preamp that is
distorting. In this case, careful
metering can help only as a
guide to the problem, as PFL
(Pre Fade Listen) metering is
generally post the EQ section
and the channel insert point,
narrowing the problem down
only to these areas. Once this
distortion is within the signal it
cannot be resolved by reducing
the overall fader volume on the
channel the gain on the mic
preamp has to be set correctly
in the first place.

From real to reel

The most critical time for
ensuring optimal gain structure
is during the recording process
itself. When youre working with
analogue equipment there is
always a certain amount of
headroom available before
distortion becomes apparent

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The TL Audio VP1 channel strip includes a high-quality mic preamp, EQ and
compressor along with an optional A/D converter.

to the extent that most true

analogue mixers have an
additional amount of headroom
(around 6dB) beyond the final
+12dB VU mark on the meter
before distortion takes place
within the output stage of the
desk itself.
When moving from the
analogue to the digital world (as
is necessary when recording
takes place) you no longer have
the luxury of this headroom;
there is a definite point of no
return, where all the bits
become one and beyond which
the signal is corrupted.
To accommodate this, digital
recorders and soundcards are
configured with a false internal
headroom, so they have to be
driven quite hard from the mixer
to actually create digital
distortion. The result is that it
can often be quite difficult to get
a decent signal level into a

that to reach 0dBFS, the mixer

could now be running towards
its own internal distortion
threshold the reasoning being
that minor analogue distortions
are far less of a problem than
any digital distortion.
One workaround is to adjust
the sensitivity settings on the
inputs of the soundcard. Instead
of matching the output of the
mixing desk to the input of the
card, it is possible to configure
the card so that it is set up with
-10dBV sensitivity. Feeding this
with the +4dBu outputs from the
desk means that 0dBFS can be
achieved at the digital inputs, at
around +4dB VU on the mixer.

Feel the squeeze

Making the most of the signal-tonoise ratio of a digital input is not
necessarily as straightforward.
With a 16-bit system, for
example, leaving 6dB of

The metering in Pro Tools enables you to see the internal levels of the mix busses.

edge off the transient peaks and

increase the level of the quieter
signals using make-up gain. It is
while recording quieter signals
that the converters suffer due to
a lack of resolution and
problems with quantisation noise
become apparent. The
advantage to compressing the
signal before the A/D process is
that it is making far better use of
the total number of bits, giving a
better signal integrity once in the
digital domain. The disadvantage
is that this compression cannot
be removed from the recorded
signal, and that the make-up
gain will also raise the noise floor

The most critical time for ensuring optimal

gain structure is during recording itself.
soundcard as an analogue desk
may have to be outputting
+12dB more than you would
expect before reaching 0dBFS
on the digital meter. The irony is

headroom reduces the signal-tonoise ratio to 90dB. One option

is to use some gentle analogue
compression on the way into the
soundcard to take some of the

Gain

Aux send

Reverb

0dB

0dB

0dB

Reverb

0dB

Channel faders
Feeding a reverb unit with a satisfactory input level will reduce the overall noise floor.
This is illustrated here by the right-hand option.

within the analogue signal,

potentially highlighting the
frailties of the preamp stage.

Digital dynamics
Once within the digital domain,
incorrect gain structure can have
a far more catastrophic effect
than the equivalent problem on
an analogue desk. Overdriving
any virtual channel beyond
0dBFS causes damage to the
audio signal. The advantage to
virtual mixers is the metering,
which means it is very easy to
trace and resolve problems with
gain structure. Every channel
be it auxiliary, buss, group or
main output has a peak meter
that provides additional visual
information about the audio
signals themselves.

When working with a busy

mix, the Peak Hold functionality
can be a real benefit as it makes
it possible to search through the
channels to see which have
overstepped over the mark and
resolve the problem. MTM

Tech terms
Unity gain

T"e %oi(t *a,e**ed as 0d0 o( a

fader3 4"ere t"e sig(a* is (eit"er
,ei(g a6%*ified or atte(uated.
Headroom

T"e a6ou(t of e*e9tro(i9 :s%a9e;

t"at is a<ai*a,*e 4it"i( a sig(a*
%ro9essor ,et4ee( its (or6a*
o%erati(g *e<e* a(d t"e %oi(t at
4"i9" distortio( 4i** o99ur.
Quantisation noise

T"e ra(do6 (oise sig(a*s t"at are

added to a( audio sig(a* duri(g
t"e %ro9ess of digitisatio(3 4"e(
t"e 6easured <a*ue of a(
a(a*ogue sig(a* is 6a%%ed o(to
t"e e=ui<a*e(t digita* ,it <a*ue.
T"is (oise ,e9o6es a %ro,*e6
o(*> 4"e( re9ordi(g <er> =uiet
audio sig(a*s.
dBFS

T"e de9i,e* ra(ge used for digita*

6eteri(g3 4"ere 0d0?@ is t"e
6aAi6u6 sig(a* *e<e* a(d a** ot"er
sig(a*s are 6easured (egati<e*>
fro6 t"at %oi(t.

F"R$%ER ()FO
Setting the gain structure for a live
performance is addressed in the
following two sites, the first more
theory-based than the second:
www.mmproductions.co.uk/gain1.html
www.prosoundweb.com/install/
tech_corner/elwl.php
Many mixer manufacturers provide
some helpful pdf files about setting up
good gain structure using their
mixers. This is Mackies link:
www.mackie.com/pdf/
CMRefGuide/Tips_Ch4.pdf

MusicTech MAGAZINE March 2007 39