ANTARES AVP-1 Owner’s Manual

AVP-1

Antares V ocal Producer

Owner’s Manual

©2002 Antares Audio Technologies. All rights reserved.
All trademarks are the property of their respective owners.
All names of microphone manufacturers and microphone model

designations appearing in this manual are used solely to identify the
microphones analyzed in the development of our digital models and
do not in any way imply any association with or endorsement by any
of the named manufacturers.

Antares Audio Technologies
231 Technology Circle
Scotts Valley, California 95066 USA

voice: [phone]
fax: [phone]
email: [e-mail]
web: www.antarestech.com

Printed in USA Rev 1.2 01/2002

ii

Contents

Getting Started

Welcome v
Technical Support vi

Introducing the Antares Chapter 1
Vocal Producer

AVP Overview 2
Auto-Tune Pitch Correction 3
Antares Microphone Modeling 7
Understanding Compression 8
What is a De-Esser? 15
Equalization 15

Setting Up Chapter 2

Setting up the AVP is easy 20

Panel Controls Chapter 3
and Connectors

The front panel 21
The back panel 24

Operation Chapter 4

Live or mixdown? 25
Patching the AVP into your system 25
Controls and Display Screens 28

Master Module 28
Microphone Modeler Module 40
Auto-Tune Module 45
Compressor/Gate Module 48
De-Esser Module 51
Equalizer/Output Module 53

iii

Get Creative Chapter 5 58

Appendix

Factory Presets 60
Realistic Mic Modeling Expectations 64

Specifications 66

Index 68

iv

Welcome!

On behalf of everyone at Antares Audio Technologies, we’d like to
offer both our thanks and congratulations on your decision to
purchase the Antares Vocal Producer.

Before you proceed any farther, we’d like to strongly encourage you
to fill out and return the AVP-1 registration card. To make it as easy
as possible, we’ve included a sticker with your serial number already
attached to the card. It’s probably a good idea also to write it in
your manual for future reference.

As an AVP-1 owner, you are entitled to receive notification of any
software upgrades, technical support, and advance announcements
of upcoming products. But we can’t send you stuff unless we know
who and where you are. So please, send that card in.

At Antares, we are committed to excellence in quality, customer
service, and technological innovation. With your purchase of the
AVP-1, you have created a relationship with Antares which we hope
will be long and gratifying. Let us know what you think. You can
count on us to listen.

Again, thanks.
The Whole Antares Crew

v

Technical Support

In the unlikely event that you experience a problem using your
Antares Vocal Producer, try the following:

1. Make another quick scan through this manual. Who knows? You
may have stumbled onto some feature that you didn’t notice the
first time through.

2. Check our web page for tips, techniques, or any late-breaking
information: www.antarestech.com

3. Call your local Antares dealer.

4. Call us at [phone] Monday through Friday between
9am and 5pm USA Pacific Standard Time.

5. Email us at: [e-mail]

For options 3, 4 and 5, please be prepared to provide the serial
number of your Vocal Producer.

vi

Chapter 1:
Introducing the Antar es Vocal Pr oducer

How to use this manual

The Antares Vocal Producer (henceforth referred to as the AVP) has a very
friendly user-interface and is extraordinarily easy to use. However, to get
the full benefit of its capabilities, we recommend that you give this
manual at least a quick once over.

If the AVP is your first experience with vocal signal processing, you’ll find a
brief introduction to the theory and application of the various processing
modules in this chapter. (More in-depth information can be found in a
variety of books on recording technique and periodically in recording­oriented magazines like Electronic Musician, EQ, Mix, Recording,and
Home Recording, among others.)

If you’re already familiar with the functions and uses of basic studio signal
processors (compressor, gate, de-esser, EQ, etc.), you can go straight to
Chapter 4 to see how they are implemented in the AVP. On the other
hand, unless you have experience with Auto-Tune and Antares Micro­phone Modeler, it’s probably wise to at least read the background infor­mation on those features in Chapter 1.

The contents of this manual

Chapter 1: Introducing the Antares Vocal Producer

The chapter you are reading. It provides an overview of the AVP as well as
background information on Auto-Tune pitch correction and Antares
Microphone Modeling. It also includes an introduction to basic concepts in
compression, expansion, gating, de-essing, and parametric equalization.

Chapter 2: Setting Up the Antares Vocal Producer

How to get the AVP up and running.

Chapter 3: Controls and Connectors

This chapter provides a reference for all of the controls, displays and
connectors on the AVP’s front and back panels.

1

Chapter 4: Operation

This is a guide to all of the features and functions of the AVP. If you’re
only going to read one chapter, this is the one.

Chapter 5: Creative Applications for the AVP

Some cool, but not-so-obvious stuff you can do with the AVP.

Antares Vocal Producer Overview

The heart of any great song is a great vocal sound. With the Antares Vocal
Producer, we’ve combined our world-renowned Auto-Tune Pitch Correc­tion and TEC-Award-winning Microphone Modeler technologies with
state-of-the-art vocal processing modules to give you everything you need
to create stunning vocal tracks in any musical style.

Live or in the studio, the AVP lets you instantly select from a library of
sounds. From gorgeously mellow to seriously twisted, we’ve included
factory presets for a wide variety of vocal styles as well as an interface that
makes it easy to create your own signature sounds. (And given the power
and flexibility of the AVP’s processing modules, we’ve even included a
selection of presets for instrumental and percussion tracks.)

The Antares Vocal Producer features:

• Auto-Tune Real-time Pitch Correction Antares’s world-renowned Auto-
Tune technology lets you correct the pitch of vocals (or solo instru­ments), in real time, without distortion or artifacts, while preserving all
of the expressive nuance of the original performance.

• Antares Microphone Modeling Our TEC Awarding-winning Microphone
Modeler technology lets you give your vocal tracks the characteristics of
a variety of high-end studio mics as well as adjust the proximity effect
associated with mic distance.

• Analog Tube Modeling Gives your vocals the warmth of a classic tube
preamp.

• Variable Knee Compressor A state-of-the-art dynamics processor with threshold, ratio, attack and decay controls as well as a continuously variable knee characteristic.

• Downward Expanding Gate The AVP’s gate, with threshold and ratio
controls, works independently of the compressor to eliminate noise and
breath sounds.

• Variable Frequency De-Esser The AVP’s de-esser tames vocal sibilance
with threshold, ratio, attack and decay controls as well as a variable
highpass frequency to match any vocal performance.

2

• Flexible Parametric EQ You can fine-tune your vocal sound with two independent bands of equalization that let you select from 6dB or 12dB high or low cut, high or low shelving with variable slope, bandpass, notch and fully parametric peaking.

• Automatic Mono or Stereo Double Tracking You can automatically mix
a doubled track into the AVP’s main output or route it to a separate
output for post-processing and mixing.

• Fully Programmable Once you’ve created the perfect vocal sound for a
particular track, every parameter can be saved as a preset for instant
recall.

• Factory Presets for a Wide Variety of Vocal Styles The AVP comes out-
of-the-box with an extensive collection of factory presets for a variety
of vocal styles. (We’ve even included a selection of presets for instru­mental and percussion tracks.)

• MIDI Automation Every variable module parameter can be controlled
via MIDI continuous controllers for realtime automation.

• Really Easy To Use No scrolling though endless menus to find the
parameter you want. Virtually every major function is only a single
button press away.

Read on for the details.

Auto-Tune Pitch Correction

In 1997, Antares first introduced the ground-breaking Auto-Tune Pitch
Correcting Plug-In for ProTools™ (eventually followed by most other plug­in formats). Here was a tool that actually corrected the pitch of vocals and
other solo instruments, in real time, without distortion or artifacts, while
preserving all of the expressive nuance of the original performance.
Recording Magazine called Auto-Tune a “Holy Grail of recording.” And
went on to say, “Bottom line, Auto-Tune is amazing… Everyone with a
Mac should have this program.” In fact, we know of quite a few people
who bought kilo-buck ProTools systems just to be able to run Auto-Tune.

The AVP’s Auto-Tune module is a hardware implementation of our Auto­Tune pitch correcting software. Like Auto-Tune, the AVP employs state-of­the-art digital signal processing algorithms (many, interestingly enough,
drawn from the geophysical industry) to continuously detect the pitch of a
periodic input signal (typically a solo voice or instrument) and instantly
and seamlessly change it to a desired pitch (defined by any of a number of
user-programmable scales).

3

A little bit about pitch

Pitch is typically associated with our perception of the “highness” or
“lowness” of a particular sound. Our perception of pitch ranges from the
very general (the high pitch of hissing steam, the low pitch of the rumble
of an earthquake) to the very specific (the exact pitch of a solo singer or
violinist). There is, of course, a wide range of variation in the middle. A
symphony orchestra playing a scale in unison, for example, results in an
extremely complex waveform, yet you are still able to easily sense the
pitch.

The vocalists and the solo instruments that the AVP is designed to process
have a very clearly defined quality of pitch. The sound-generating mecha­nism of these sources is a vibrating element (vocal chords, a string, an air
column, etc.). The sound that is thus generated can be graphically repre­sented as a waveform (a graph of the sound’s pressure over time) that is
periodic. This means that each cycle of waveform repeats itself fairly
exactly, as in the periodic waveform shown in the diagram below:

Because of its periodic nature, this sound’s pitch can be easily identified
and processed by the AVP.

Other sounds are more complex. This waveform:

is of a violin section playing a single tone. Our ears still sense a specific
pitch, but the waveform does not repeat itself. This waveform is a summa­tion of a number of individually periodic violins. The summation is non­periodic because the individual violins are slightly out of tune with respect
to one another. Because of this lack of periodicity, Auto-Tune would not
be able to process this sound.

Some pitch terminology

The pitch of a periodic waveform is defined as the number of times the
periodic element repeats in one second. This is measured in Hertz (abbre­viated Hz.). For example, the pitch of A3 (the A above middle C on a
piano) is traditionally 440Hz (although that standard varies by a few Hz. in
various parts of the world).

4

Pitches are often described relative to one another as intervals, or ratios of
frequency. For example, two pitches are said to be one octave apart if
their frequencies differ by a factor of two. Pitch ratios are measured in
units called cents. There are 1200 cents per octave. For example, two tones
that are 2400 cents apart are two octaves apart. The traditional twelve­tone Equal Tempered Scale that is used (or rather approximated) in 99.9%
of all Western tonal music consists of tones that are, by definition, 100
cents apart. This interval of 100 cents is called a semitone.

How Auto-Tune detects pitch

In order for Auto-Tune to automatically correct pitch, it must first detect
the pitch of the input sound. Calculating the pitch of a periodic waveform
is a straighforward process. Simply measure the time between repetitions
of the waveform. Divide this time into one, and you have the frequency in
Hertz. The AVP does exactly this: It looks for a periodically repeating
waveform and calculates the time interval between repetitions.

The pitch detection algorithm in the AVP is virtually instantaneous. It can
recognize the repetition in a periodic sound within a few cycles. This
usually occurs before the sound has sufficient amplitude to be heard. Used
in combination with a slight processing delay (no greater than 4 millisec­onds), the output pitch can be detected and corrected without artifacts in
a seamless and continuous fashion.

The AVP was designed to detect and correct pitches up to the pitch C6. If
the input pitch is higher than C6, the AVP will often interpret the pitch an
octave lower. This is because it interprets a two cycle repetition as a one
cycle repetition. On the low end, the AVP will detect pitches as low as 42
Hz. This range of pitches allows intonation correction to be performed on
all vocals and almost all instruments.

Of course, the AVP will not detect pitch when the input waveform is not
periodic. As demonstrated above, the AVP will fail to tune up even a
unison violin section. But this can also occasionally be a problem with solo
voice and solo instruments as well. Consider, for example, an exceptionally
breathy voice, or a voice recorded in an unavoidably noisy environment.
The added signal is non-periodic, and the AVP will have difficulty deter­mining the pitch of the composite (voice + noise) sound. Luckily, there is a
control (the Sensitivity control, discussed in Chapter 4) that will let the
AVP be a bit more casual about what it considers “periodic.” Experiment­ing with this setting will often allow the AVP to track even noisy signals.

5

How Auto-Tune corrects pitch

Auto-Tune works by continuously tracking the pitch of an input sound and
comparing it to a user-defined scale. The scale tone closest to the input is
continuously identified. If the input pitch exactly matches the scale tone,
no correction is applied. If the input pitch varies from the desired scale
pitch, an output pitch is generated which is closer to the scale tone than
the input pitch. (The exact amount of correction is controlled by the Speed
parameter, described below and in Chapter 4.)

Scales

The heart of Auto-Tune pitch correction is the Scale. The AVP comes with
25 preprogrammed scales. For each Scale you can define which notes will
sound and which won’t. And for each note that will sound, you can decide
whether the AVP will apply pitch correction to input pitches near that
note or leave those pitches uncorrected.

You can also edit any of the preprogrammed scales and save your custom
scale as part of a Preset.

Speed

You also have control over how rapidly, in time, the pitch adjustment is
made toward the scale tone. This is set with the Speed control (see Chap­ter 4 for more details).

• Fast Speed settings are more appropriate for short duration notes and
for mechanical instruments, like an oboe or clarinet, whose pitch
typically changes almost instantly. A fast enough setting will also
minimize or completely remove a vibrato. At the fastest setting, you
will produce the now-infamous “Cher effect.”

• Slow Speed settings, on the other hand, are appropriate for longer
notes where you want expressive pitch gestures (like vibrato) to come
through at the output and for vocal and instrumental styles that are
typified by gradual slides (portamento) between pitches. An appropri­ately selected slow setting can leave a vibrato unmodified while the
average pitch is accurately adjusted to be in tune.

6

An example

As an example, consider this before-and-after graphic representation of
the pitch of a vocal phrase that contains both vibrato and expressive
gestures.

CORRECTED
BY AVP

D3



C

3

B2

10.0 10.5 11.0

In the original performance, we can see that although the final note
should be centered around D, the vocalist allowed the tail of the note to
fall nearly three semitones flat. The “after” plot is the result of passing
this phrase through the AVP set to a D Major Scale (with C# and B set to
”Blank”) and a Speed setting of 10. That Speed causes the pitch center to
be moved to D, while still retaining the vibrato and expressive gestures.
(Setting C# and B to ”Blank” is necessary to keep the AVP from trying to
correct the seriously flat tail of the last note to those pitches. See Chapter
4 for more details.)

ORIGINAL
PERFORMANCE

Antares Microphone Modeling

If you’ve spent any time lately flipping through the pages of pro audio
magazines, you have almost certainly noticed the intense focus on micro­phones. From the proliferation of exotic new mics to the almost cult-like
following of certain historical classics, never has the choice been greater.
But amassing a substantial collection of high-end mics is financially pro­hibitive for all but the most well-heeled studios.

Now, using our patented Spectral Shaping Tool™ technology, we’ve
created digital models of a variety of microphones. Simply tell the AVP
what type of microphone you are actually using and what type of micro­phone you’d like it to sound like. It’s as simple as that.

7

With the AVP, you can record each track through a model of the type of
mic that will best produce that ideal sound you’re looking for. Or use it in
live performance to get the sound of mics you’d never consider using on
stage. You can even use it during mixdown to effectively change the mic
on an already recorded track. And for that final touch of perfection, you
can even add some tasty tube saturation.

About the technology

The models employed by the AVP are not derived from theoretical consid­erations. They are generated by a proprietary analysis process that is
applied to each physical mic modeled. Not only the sonic characteristics,
but the behavior of other parameters such as low-cut filters or proximity
effects accurately reflect the specific performance of each microphone we
model.

Another advantage of our model-based approach is that there is essen­tially no processing delay apart from the natural phase effects of the
microphones being modeled.

Finally, the quality and signal-to-noise characteristics of the processing are
pristine. Because of our commitment to model-based processing, there are
none of the limitations or distortions characteristic of FFT-based algo­rithms. The quality of the output is limited only by the quality of the
input.

So what exactly does it do?

While there is a lot of fairly complicated stuff going on under the hood,
the essential functionality of the AVP’s Mic Modeling module is really
quite simple. Basically, audio originally recorded by a microphone is input
to the AVP where it is first processed by a “Source Model” which serves to
neutralize the known characteristics of the input mic. The audio is then
processed by a second “Modeled Mic” model which imposes the character­istics of the modeled mic onto the previously neutralized signal. Finally,
the audio is passed through a model of a high-quality tube preamp
offering the option of classic tube saturation distortion.

Understanding Compression

Compression is probably the most widely used (and potentially confusing)
signal process used in today’s studios. Simply put, compression reduces the
dynamic range of a signal. That is, it reduces the difference in loudness
between the loudest and quietest parts of a piece of music. Another way
to think about this is that the compressor is acting as an automatic fader
which fades down when the signal gets loud and fades back up when the
signal gets soft.

8

Why reduce the dynamic range? Consider the problem of mixing the vocal
in a contemporary rock or pop song. Typically, pop music has a relatively
consistent level of loudness. If an uncompressed vocal track is added to a
typical pop mix, loudly sung words or syllables would jump out of the mix,
while quieter phrases would be buried beneath the instrumental texture.
This is because the difference between the loudest and softest sounds in
the vocal - its dynamic range - is very large. This same problem occurs for
any instrument which has a dynamic range larger than the music bed into
which it is being mixed. (For that reason, most instruments, not just vocals,
undergo some compression in the typical mix.)

By using a compressor to decrease the dynamic range of the vocal, the
softer sounds are increased in loudness and the loudest sounds are re­duced in loudness, tending to even out the overall level of the track. The
overall level of the compressed track can then be increased (using what is
referred to as “make-up gain”), making the vocal track louder and more
consistent in level, and therefore easier to hear in the mix.

Threshold and Ratio

How is compression measured? What is a little compression and what is a
lot of compression?

The effect a compressor has on a track is determined by the settings of its
threshold and ratio. The threshold is the level above which the signal is
attenuated. The ratio is the measure of how much the dynamic range is
compressed.

The graph shown below shows the relationship between the input level of
a signal and the output level of the signal after compression. Notice that
signals that are louder than the threshold are compressed (reduced in
level) while those softer than the threshold are unchanged.

As the input signal exceeds the threshold, gain reduction (reduction in
loudness) is applied. The amount of gain reduction that is applied depends
on the compression ratio. The higher the compression ratio, the more gain
reduction is applied to the signal.

The graph shows the relationship between compression ratio and gain
reduction. Examine the 2 to 1 ratio curve. For signals above the threshold,
this setting transforms a range of loudness 2 units large into a range of
loudness one unit large (i.e., if the input signal gets “x” units louder, the
compressed signal increases by only “x/2” units).

9

1 TO 1 RATIO

OUTPUT

LEVEL

LOUDER

THRESHOLD

I/O CURVE

LOUDER

INPUT LEVEL

2 TO 1 RATIO

4 TO 1 RATIO
8 TO 1 RATIO
99 TO 1 RATIO

Limiting

Examine the 99:1 curve in the above graph. This setting reduces all sounds
above the threshold to the same loudness. This is called limiting. Limiting
is usually employed to allow a dynamic signal to be recorded at a maxi­mum level with no risk that transient peaks will result in overload. In this
application, the threshold setting (usually set relatively high) determines
the extent to which the peaks will be limited.

Dynamic Expansion and Gating

Sometimes, it is desirable to increase the difference between the quietest
signal and the noise in a recording by using a downward expander. A
typical application would be eliminating room noises and breath sounds
that can be heard between the phrases of a recorded vocal part.

10

The graph below shows the curveÉ<or a downward expander . Notice that
above the threshold, the curve follows a 1 to 1 ratio (i.e., is unaffected by
the gate). For each unit of input change below the threshold the output
changes by two units. This is called a 1 to 2 expansion ratio.

As the input signal drops below the threshold, its output level drops at
twice the rate it would using a 1 to 1 ratio. In effect, sounds below the
expander threshold are “faded out” more quickly than they would be
normally.

OUTPUT

LEVEL

1 TO 1 RATIO

LOUDER

THRESHOLD

1 TO 2 EXPANSION RATIO

LOUDER

INPUT LEVEL

When expanders use ratios higher than 1:10, sounds below the threshold
are faded out very rapidly. This effect is called gating and can sound very
abrupt. Adjusting the gate ratio can smooth out the abrupt change. The
graph below shows the input/output curve for a typical gate.

1 TO 1 RATIO

OUTPUT

LEVEL

LOUDER

THRESHOLD

1 TO 99 EXPANSION RATIO

LOUDER

INPUT LEVEL

11

Sounds that are louder than the threshold get “through the gate” un­changed. Sounds that are below the threshold are not heard. Gates can be
used to great effect in processing drum tracks where sounds from the
other instruments in the drum set leak through the mike of the instru­ment being recorded. Gates are also used frequently to “gate off” a
reverb tail or the ringing from an insufficiently damped drum head.

Compression and Expansion Combined

The AVP allows you to use both compression and expansion simulta­neously. This ability is useful in taming the typical problems that arise
when processing vocal tracks. The graph below illustrates the use of
compression with a downward expanding gate.

OUTPUT

LEVEL

COMPRESSOR THRESHOLD

GATE THRESHOLD

1 TO 99 EXPANSION RATIO

LOUDER

INPUT LEVEL

4 TO 1 RATIO

Using this setting, levels above the compressor threshold will be com­pressed at a 4 to 1 ratio. Levels below the compressor threshold but above
the gate threshold will not be changed. Levels below the gate threshold
will be gated out completely.

Used on a vocal track, this setting will compress only hot peaks in the
voice, while gating out the room sounds, mike stand sounds, and breath
noises in the track. Precisely what gets compressed and gated is a function
of the compressor and gate threshold settings.

12

The graph below shows a dynamic expander. In this application, the gate
threshold and ratio are set to gently expand the program material at a 1.5
to 1 ratio. The compressor ratio is set to 1 to 1. The setting is useful for
repairing over-compressed material or for adding some punch to drums or
other percussive sounds.

OUTPUT

LEVEL

COMPRESSOR
THRESHOLD

GATE THRESHOLD

LOUDER

1 TO 5 EXPANSION RATIO

LOUDER

INPUT LEVEL

Hard Knee/Soft Knee

The graphs shown above have what are described as “hard knees” in their
gain curves. This means that as the signal passes through the threshold,
the gain reduction it receives will begin abruptly. In settings where the
compression or expansion ratios have high values, the abrupt change can
be heard and often sounds artificial.

OUTPUT

LEVEL

To make it possible to create settings where the dynamic effects are more
natural sounding, the AVP incorporates a Knee control which allows you
to soften the transition between sections of the gain curve. The graph
below shows a curve which has “soft knees,”making the dynamic transi­tions more subtle.

COMPRESSOR THRESHOLD

SOFT KNEES
KNEE = 100

GATE THRESHOLD

INPUT LEVEL

13

Attack and Release Times

The attack time of a compressor is how long it takes for the compressor to
react once the input level has met or exceeded the threshold level. With a
fast attack time, the signal is brought under control almost immediately,
whereas a slower attack time will allow the start of a transient or a
percussive sound to pass through uncompressed before the processor
begins to react.

For sounds without percussive attacks (voices, synth pads, etc.), a fairly
short attack time is usually used to ensure even compression. For instru­ments with percussive attacks (drums and guitars, for example), a slower
attack time is typically used to preserve the attack transients and, hence,
the characteristic nature of the instruments.

The illustration below shows the effect of various the attack times.

14

UNCOMPRESSED INPUT COMPRESSED

1 mSEC ATTACK

COMPRESSED
10 mSEC ATTACK

The release time of a compressor is the time it takes for the gain to return
to normal after the input level drops below the threshold. A fast release
time is used on rapidly varying signals to avoid affecting subsequent
transients. However, setting too quick a release time can cause undesirable
artifacts with some signals. On the other hand, while slower release times
can give a smoother effect, if the release time is too long, the compressor
will not accurately track level changes in the input. Slow release times may
also result in audible level changes known as “pumping.”

What is a De-Esser?

When recording spoken or sung material, the sibilants (Ss, Ts, CHs, and
SHs) in the track often sound louder than the rest of the signal. The effect
is unnatural and often irritating. The solution to this problem is to com­press only the sibilants, thereby lowering their level relative to the rest of
the track. Processing a signal this way is called de-essing.

The diagram below shows how analog hardware is traditionally config­ured to accomplish de-essing.

Equalization

IN

IN OUT

COMPRESSOR

HIGH
PASS

FILTER

SIDECHAIN
INPUT

OUT

Only the sibilants pass through the highpass filter. When the input signal
contains sibilant material, the output of the filter causes the compressor
to compress the signal. The compressor only operates when a sibilant is
present.

The AVP uses a digital algorithm to implement the de-esser function.
While the details of the algorithm are quite complex, the resulting effect
is functionally equivalent to the diagram above.

The AVP’s two bands of equalization each offer seven different filter
types: Low Pass (6dB/octave and 12dB/octave), Low Shelf, Band Pass,
Notch, Peaking, High Shelf, and High Pass (6dB/octave and 12dB/octave).
Each filter type has its own characteristics and applications. The graphs
used in the next section show the frequency response for each type with
the settings used to generate the curves notated next to the graph.

15

Low Pass - High Pass Filters

The low pass and high pass filters available in the AVP offer both a 6dB
per octave and a 12dB per octave roll-off characteristic. The 6dB per
octave versions offer a more subtle effect, while the 12dB per octave
roll-off is useful for attenuating sub-sonic noise, rumble, mic stand noise,
high frequency hiss, and other environmental noises encountered in the
recording process. Additionally, the 12dB per octave versions provide a
“Q” control that allows you to create a variable height peak at the cut-off
frequency.

LOG

MAGNITUDE

(dB)

LOG

MAGNITUDE

(dB)

6

0

-6

-12

-18
100 300 1000 3000 10000 22050

50

6

0

-6

-12

FREQUENCY

LP

LOW PASS FILTER
Frequency: 1,000 Hz
Gain: N/A
Bandwidth: N/A

HP

HIGH PASS FILTER
Frequency: 1,000 Hz
Gain: N/A
Bandwidth: N/A

16

-18
100 300 1000 3000 10000 22050

50

FREQUENCY

Shelving Filters

Shelving filters are used primarily as “tone controls,” cutting or boosting
whole regions of the spectrum. (You can think of them as fancy versions of
the traditional “Bass” and “Treble” controls you’d find on home stereos or
boom boxes.) A high shelf filter, for instance, acts by raising or lowering
the part of the spectrum above the cut-off frequency.

The graphs below show the response of the high shelf and low shelf filters
at +12dB gain. Notice that the slope of the roll-off is 6dB per octave. The
AVP’s shelf filters provide a slope control that let’s you vary the filter’s
slope between 2dB and 12dB per octave.

LOG

MAGNITUDE

(dB)

LOG

MAGNITUDE

(dB)

6

0

-6

-12

-18
50

100 300 1000 3000 10000 22050

FREQUENCY

18

12

6

0

HS

HIGH SHELF FILTER
Frequency: 1,000 Hz
Gain: +12 dB
Bandwidth: N/A

LS

LOW SHELF FILTER
Frequency: 1,000 Hz
Gain: +12 dB
Bandwidth: N/A

-6
50

100 300 1000 3000 10000 22050

FREQUENCY

17

Peaking Filter

The peaking filter is the traditional fully parametric EQ. It can be used
to subtly accentuate or attenuate a frequency or for much more radical
effects.

In the AVP, the peaking filter works over a range of 20 Hz to 20 kHz and
can boost or cut the signal at the selected frequency by ± 18dB. Addition­ally, you can vary the bandwidth from 0.1 to 4.0 octaves.

The graphs below show the effect of changing the bandwidth control of
the peaking filter.

LOG

MAGNITUDE

(dB)

LOG

MAGNITUDE

(dB)

18

12

6

0

-6
50

100 300 1000 3000 10000 22050

FREQUENCY

18

12

6

0

BP1

PEAKING FILTER
Frequency: 1,000 Hz
Gain: +12 dB
Bandwidth: 1.0 octave

BP1

PEAKING FILTER
Frequency: 1,000 Hz
Gain: +12 dB
Bandwidth: 0.1 octave

18

-6

100 300 1000 3000 10000 22050

50

FREQUENCY