As a practical matter—
the CALM Act or to
to a specific loudness
are relying more on
loudness processors to
automatically make adjustments to the audio
There are two broad categories of loudness
processors: real-time and file-based.
Within each type, loudness processing can
be a stand-alone function or incorporated
as part of a total processing package.
Real-time loudness processors operate
in nearly real time (with some buffering
and processing delay) to meter loudness
of an incoming signal (mono or multichannel)
and then typically continuously make
audio level adjustments depending on the
rules or presets it is given.
|Fig. 1: General signal flow of a multiband real-time loudness processor, Linear Acoustic Aeromax
A file-based loudness processor, on the
other hand, analyzes loudness from audio
that was recorded as a digital file, typically as
part of a video file. The analysis occurs over
the entire length of an audio piece and then
a scaling factor (gain or attenuation), if needed,
is applied to the entire content, based on
preset rules, so that the audio output is delivered
at that new (target)
While they can be used
for treating loudness on
archival material on tape
(instead of in a digital format),
are typically installed in the
last stage of the audio signal
chain, before an encoder, to
catch any loudness problems
that would make the
A real-time processor
fixes loudness “by adjusting
dynamic range,” said Tim Carroll, Telos
Alliance CTO and Linear Acoustic founder.
The processor continuously measures
program loudness of the input audio signal
according to the ITU-R BS.1770 standard
(currently version 3, from August 2012).
Then automatic gain control, or in other
words, compression, is applied to adjust the
level to meet the target. Think of this as riding
gain with an audio fader.
|Fig. 2: Comparison of multiband (left) and wideband multiloop (right) block diagrams for real-time
“The [processor] is constantly adjusting
the level,” said Peter Pörs, managing director,
Jünger Audio GmbH. “The fader is never
in a fixed state. It’s moving so slowly that
you don’t perceive it.”
For content at relatively controlled levels,
adjusting the AGC too fast will be audible,
yet if a loud transient occurs, the processor
must be able to react quickly to pull
it down. This is why processors typically
have an output limiting stage.
WIDEBAND AND MULTIBAND
Processors differ in how the AGC is applied.
Some apply AGC across the entire
audio spectrum (wideband). Others use
the multiband approach where they break
up the full audio spectrum into sections or
bands and apply AGC individually to each
band. Different attack and release times can
be set for each band.
According to Carroll, if the processor
runs in straight wideband this, in general,
can make processing adjustments more
audible. As an example, a loud thump
could bring down the level of a whole
program, even though it’s only the lower
frequencies that caused the level to spike.
With multiband processing in this scenario,
only the lower frequencies would be
reduced, (Fig. 1).
Four or five bands generally are adequate
for multiband loudness processors, according
to Carroll, and this is what is typical.
Another idea behind multiband processing
is the way us humans perceive sound.
We don’t hear linearly across the audible
frequency range. We are more sensitive to
mid-range sounds compared with those of
higher and lower frequencies, but the difference
changes as the audio level changes.
For example, for normal hearing at
low audio levels, a sound at 100 Hz must
be raised about 15 dB higher than one at
1,000 Hz for the two tones to be perceived
as equally loud. As audio levels increase, the
lower frequencies don’t need to be raised
quite that much compared to the mid-range
to be perceived as equally loud.
Two researchers from Bell Labs, Harvey
C. Fletcher and Wilden A. Munson, studied
this phenomenon and in the early 1930s
published their results with graphs of equal
loudness curves across the audio spectrum
and at different audio levels. These have
come to be known, not surprisingly, as the
|Fig. 3: Block diagram of Jünger Audio Level Magic loudness management
That’s why if you compress the entire
signal you change the relative levels of the
high and low frequencies to the mid-range,
and that, according to Bob Nicholas, director
of international business development
for Cobalt Digital, changes the character of
the sound. This can have a negative effect
“Multiband is not trying to keep things
spectrally flat, but to keep things spectrally
balanced,” Carroll said.
Nicholas said that multiband AGC is
more applicable to a sound source that’s
a mix of different signals and wideband
is more for single source signals, like that
used on a channel strip of an audio console.
Taking a different tack, Pörs said that the
multiband approach can produce anomalies
when the different frequency bands are
summed together. “The possible difficulty is
that [with] the overlapping zones of the
filters, a precise summation of the signals
is nearly impossible [and that] leads to coloration,”
he said. “That’s why we came to
wideband.” (See Fig. 2.)
Wideband with a twist, that is. “We have
a different processing design approach,”
Pörs said. “We call it multiloop design.” This
design incorporates a series of gain controls,
with each “fader” controlled separately.
(See Fig. 3.)
“The various loops each work over
the entire frequency spectrum,” Pörs said.
“They work in parallel, each with a different
set of attack and release parameters.
Each loop develops a control signal which
is then summed with the controls from
the other loops to produce a single gain
control signal applied to one gain control
The algorithms in the processor provide
automatic adjustment of the attack and release
time based on how the input signal
changes over time. “This is called ‘adaptive
dynamic range control,’” Pörs said. “By monitoring
the waveform of the incoming audio,
the system can set relatively long attack
times during steady-state signal conditions,
but very short attack times when there are
In addition, the Jünger multiloop design
allows for a very short time delay to be put
in the audio signal path. “This lets the gain-changing
elements ‘look ahead’ and determine
the correction needed and to apply it
to the delayed signal just in time to control
even the fastest transients,” Pörs said.
No matter what design, it must be set
and used correctly. More on this later.
Mary C. Gruszka is a systems design engineer
and consultant based in New York.
She can be reached via TV Technology.