Understanding & Emulating Vintage Effects

Sound Techniques

Published
in SOS January 2001

Printer-friendly version

Technique : Effects /
Processing

Modern digital effects
units always include emulations of analogue effects such as tape delay and
flanging -- but none of them ever seem quite like the real thing. Paul White
explains how these vintage effects worked, and offers insight into how our
modern attempts could be made more accurate.

We live in a digital age, but you can't go into a studio without
hearing stories about how good things use to sound in the old days. But were the
old electromechanical effects really that good, or have we fallen foul of the
'nostalgia isn't what it used to be' syndrome? Certainly some old effects were
noisy and unreliable, but in most cases, there was an element to the sound that
the human ear found especially pleasing in a musical context. Sometimes the
reasons are fairly obvious, but other times the magic element remains elusive,
nowhere more so than in the case of tape flanging or phasing.

Tape Phasing &
Flanging

Tape phasing, commonly known as
tape flanging, is a unique effect, and though some digital flangers have managed
to approximate it, I've yet to hear a truly convincing emulation. If you've
never experimented with tape flanging, the effect is created by running two
identical copies of the same recording on open-reel analogue recorders (usually
in mono) and then summing the two outputs together via a mixer at exactly the
same levels. The two recordings are started together -- a hit-and-miss business
at the best of times -- then the speed of one of the machines is slowed slightly
by using hand pressure on the tape reel. The idea is not to get so far behind
that you can hear a tangible ADT-style delay, but simply to produce a comb
filtering effect. Comb filtering occurs by virtue of the addition and
subtraction of frequencies that end up being in-phase or out-of-phase as
determined by the delay time. Whichever machine is leading is then slowed down
so that the delay decreases until the point where the other machine takes the
lead. As the relative delay between the two tapes changes and finally passes
through zero, the familiar whooshing effect is created as the comb filter sweeps
through different frequencies in the source material. And so it continues with
the leading machine being slowed manually so that the two recordings drift in
and out of phase with each other. Because tape flanging is literally hands on,
the effect is different every time. Some of the more sophisticated studios used
electronic speed control instead of hand braking to create the
effect.

Modern flangers seek to emulate
this effect by digitally delaying one signal relative to another (by just a few
milliseconds), then modulating the delay time using a low-frequency oscillator,
or LFO. To make the effect stronger, some of the output is fed back to the
input, which adds resonance to the comb-filtering effect. Note that with tape
flanging, no feedback is used. Artificial flanging of this kind sounds different
to tape flanging for a number of reasons -- the LFO-controlled modulation is
regular, feedback is used to add depth to the effect and the delay between the
two signals never passes through zero, as it does when two tape machines are
used.

A refinement of this method is
to delay one signal by a very small amount (say 5mS), then modulate the delay of
the other signal path so that it slowly changes from less than 5mS to more than
5mS. This provides the 'through zero' element of the effect but does nothing to
break the regularity of the modulation unless the delay time is adjusted by
hand. Furthermore, if feedback is applied, it doesn't create the desired effect,
as the feedback-induced resonance will be a function of the whole DDL delay
time, whereas the comb-filtering effect itself is related to the difference
between the two delay times. A simple setup for through-zero flanging is shown
in Figure 1 alongside the original tape-based arrangement.

In theory, it should be possible
to emulate tape flanging much more closely by using techniques such as physical
modelling. For example, one reason the effect sounds the way it does with
analogue tape machines is that analogue machines don't have the precise phase
response of a digital system. For example, put a 1kHz square wave into a digital
recorder or effects processor and what comes out will be recognisable as a
square wave. Not so with analogue tape -- the necessary frequencies are all
there, but because of phase shifts in the electronic and magnetic components of
the system, their time relationship is disturbed, which is why the waveform
looks very different to the original. The simulations might be significantly
closer if we were able to emulate this smearing before delaying the signals, as
well as introducing more randomisation into the modulation.

Plate
Reverb

During the '60s and '70s, most
artificial reverb used on recordings was generated using a reverb plate,
sometimes called (inaccurately) an echo plate.The reverb plate is an ingeniously
simple device, but it takes a lot of tweaking at the design stage to get it
sounding right. Plates work by suspending a thin sheet of metal under tension
within a rigid frame via springs or clamps attached to the corners. A transducer
similar to the voice-coil of a cone loudspeaker is used to inject audio energy
into the plate and two or more contact mics fixed to the surface of the plate
then pick up the vibrations inside it and feed them to preamps connected to the
console effect returns. By feeding the different contact mics to left and right
channels, a pseudo stereo reverb output is created.

Because metal plates have a
tendency to 'ring', getting the plate thickness, size, material and tension
right is quite an art, and some pre- and/or post-reverb EQ is invariably needed
to fine-tune the sound. Furthermore, because the plate is very sensitive to
external sounds and vibrations, it has to be mounted in a soundproof box,
ideally on shockmounts.

Unlike digital reverbs, which
have innumerable adjustable parameters, the plate reverb relies purely on EQ for
tonality and physical damping for decay-time control (usually via a motorised
felt pad). Pre-delay was often added by using an open-reel tape machine on the
input, and replaying the input signal via the replay head to exploit the time
gap between the record and replay head. You just had to hope the tape reel
didn't run out during a mix...

Because a typical plate may only
be between one and two square meters, and because sound travels much faster in
metal than in air, the reflection density within the plate builds up very
quickly following an impulse. The sound from the input transducer spreads
rapidly across the surface of the plate in all directions until it encounters
the edges of the plate, whereupon it is reflected and re-reflected back into the plate. To get the most random
reflection build-up, it's best to have the transducer mounted a little way
off-centre, and to get a wide stereo image, the two pickups are sited at
slightly different distances from the plate edge, as shown in Figure 3
(right).

The characteristic plate sound
is bright and extremely dense, with little or no impression of individual early
reflections. The reverb builds very quickly and decays smoothly with a maximum
undamped decay time of several seconds. Digital reverbs can provide a reasonable
emulation of the coloration and envelope of a plate reverb unit, but only the
more processor-intensive models produce the speed of reflection density build-up
required to be truly convincing. Plate reverb was very popular for vocal and
drum treatments, and though digital reverb has largely replaced it, many purists
still prefer the 'real' plate sound for certain applications.

Spring
Reverb

Those who couldn't afford plate
reverbs used spring reverb -- a system still used today in guitar combos. The
physical principles are similar to those of the plate reverb, except that sound
is injected into one end of a loosely coiled metal spring rather than a metal
plate, usually via a small magnetic transducer. A pickup transducer at the other
end picks up the sound as it reflects back and forth along the
spring.

Springs invariably impart a
metallic coloration to the sound and they also tend to have a cyclic
characteristic as percussive sounds cause vibrations to bounce back and forth
along the spring in a fairly regular manner. Excessive input levels cause the
springs to 'twang', so some systems incorporated an input limiter. To help even
out the coloration and cyclic modulation, it's possible to use two or more
springs operating side by side, each with slightly different mechanical
characteristics and serviced by individual transducers. Using two similar sets of springs to treat left and
right channels produces a convincing stereo effect due to the non-correlated
nature of the two spring outputs.

Like plate reverbs, their main
advantage is that they don't produce the gritty 'shattering' early-reflection
effects of digital reverberators. Figure 4 shows a stereo spring
reverb.

Tape Loop
Echo

Before digital electronics and
charge-coupled delay lines (analogue echo), delay effects were invariably
created using tape-loop echo units such as the Echoplex, Roland's famous RE201
Space Echo, or, at a somewhat lower cost, the Watkins Copicat. They all worked
on the same principle -- a loop of tape passes around a series of heads starting
with an erase head, followed by a record head fed from the signal to be treated.
Playback heads are positioned after the record head to provide the echoes, and
some of the delayed signal is fed back into the record circuitry to create
decaying echoes. Delay time is varied by switching heads or varying the tape
speed, and models with multiple heads usually have switching systems for setting
up different delay patterns (see Figure 5).

Basic digital delays only
approximate tape-loop devices, even if multiple delay taps can be set to
different delay times. There are various reasons why tape-based systems sound so
distinctive; one of the main ones is the restricted frequency response of a loop
of tape that's been dragged over a set of tape heads thousands of times. The
tube circuitry of the original models also had a limited bandwidth and
introduced a significant amount of harmonic distortion. This, combined with
tape's tendency to saturate meant that when feedback was used, successive echoes
became less bright and more distorted, creating a sense of the sound receding
into the distance. On top of that, there was instability in the tape path caused
by worn rubber pinch rollers that translated into low-level pitch modulation. As
with distortion, this type of modulation becomes cumulative when feedback is
used. Another feature that most of us would rather forget was tape noise, but
there's no denying that a good tape-echo unit has a much more 'organic' sound
than even the best digital emulations.

In conclusion, it is odd that
vintage equipment is now valued because of its sonic 'imperfections', yet none
of these were designed in deliberately. Modelling is still in its infancy,
however, and I don't expect it to be long before vintage effects can be
replicated much more accurately. Perhaps then we will be able to move on to
inventing the future rather than striving to recreate the past.

The Leslie Speaker

When the Hammond tonewheel
organ was invented, one of its limitations was that the mechanical
oscillators were all phase locked with each other due to the fact
that the tonewheels were mounted on a common shaft. This produced a
somewhat sterile sound in comparison with pipe organs, where minor
tuning errors across the numerous pipes produced a very pleasing
natural chorus effect. Vibrato was added to try to stir the sound up
a little, but it was the Leslie cabinet that really shaped the
Hammond sound.

The Leslie is effectively a
mechanical chorus device with some very interesting properties.
Essentially, the Leslie comprises two speaker systems fed from a
crossover with a small horn handling the high end and a conventional
cone driver handling the low and mid frequencies. The low-frequency
modulation effect is created by a rotating baffle system rather like
a hat box with an opening in one side. The speaker faces upwards
into this baffle, which is turned via a motor, the speed of which
can be switched from fast to slow by the organ player.

At the
high-frequency end, the modulation is created by rotating the horn
itself, and to keep the system in balance, an opposing dummy horn is
used. A simplified schematic of a Leslie cabinet is shown in Figure
2. The horn is spun by a separate motor so that the low and high
frequency modulations aren't locked together, and because of
mechanical inertia, changes in speed take place over a period of a
couple of seconds rather than being immediate.

Though this
arrangement is mechanically simple, creating a digital emulation is
more complex than it first appears. Imagine what's happening with
the low-frequency rotor first. When the aperture is facing the
listener or microphone, the sound will be at its loudest and
brightest, although the tonality will be affected by the acoustic
properties of the baffle, the speaker enclosure and any distortion
added by the amplifier, which was originally tube driven. As the
baffle turns away from the listener, the level will drop and the
frequency response will change according to the off-axis
characteristics of the baffle system. To complicate matters further,
an element of doppler shift will also be introduced as the baffle
surfaces reflecting the sound are moving relative to the listener,
although this is less pronounced for the low-frequency driver than
for the horn.

If the room
interaction is taken into account, the reflections from the moving
aperture will also be continually changing, and these will combine
with the direct sound.

A similar
thing happens with the horn, except that, as the sound source is
smaller than for the low-frequency end (the mouth of the horn), and
because it is describing an arc, the doppler effect is more
pronounced, producing a noticeable degree of pitch modulation. The
horn is also more directional than the low-frequency baffle system,
so the amplitude modulation is greater. If anything, the room
reflections contribute more significantly to the sound of the
high-frequency rotor because high-frequency reflections tend to be
more directional, the overall result being a very complex mix of
amplitude and frequency modulation combined with dynamic tonal
change and moving room reflections.

When you
come to mike up a Leslie, the effect changes again because if the
mic is close, the amplitude modulation components are emphasised
while the room reflections make less of a contribution. There's also
a certain amount of wind noise from the rotating baffle and horn
that can be heard on some recordings as a kind of 'chuffing' or
fluttering. Using a stereo pair with one mic each side of the
cabinet creates a very deep stereo effect that includes an element
of fast panning, while miking from further away reduces the degree
of amplitude modulation and increases the contributions of the room
reflections.

Electronic
rotary speaker simulators range from little more than
chorus/flangers with appropriate modulation rates and depths to more
advanced modelling where the low-frequency rotor and high-frequency
horn sounds are generated separately after first splitting the
signal into two bands using a crossover. The processing usually
involves both amplitude and frequency modulation, and because of the
characteristic overdrive sound of a valve Leslie, some form of
simulated tube overdrive is often included as well. Modelling the
dynamically changing room reflections is a rather more complex
business and is approximated only in more serious
simulators.

One of the
best Leslie simulations I've heard recently is part of Native
Instruments' B4 plug-in instrument where the user can edit a
number of separate parameters associated with the high- and
low-frequency rates, tonality and acceleration as well as overdrive
and virtual mic positions. As DSP power continues to become more
affordable, even more ambitious emulations are likely in the future.

Published
in SOS January 2001

DAW Tips from SOS

100s of great articles!

Cubase

Digital
Performer

Live

Logic

Pro
Tools

Reaper

Reason

Sonar

Home | Search
| News | Current
Issue | Tablet Mag |
Articles | Forum
| Blog |
Subscribe | Shop
| Readers Ads

Advertise
| Information |
Privacy
Policy | Support |
Login Help

Email: Contact
SOS

Telephone: +44 (0)1954 789888

Fax: +44 (0)1954 789895

Registered Office: Media House, Trafalgar Way, Bar Hill,
Cambridge, CB23 8SQ, United Kingdom.

Sound On Sound Ltd is registered in England and Wales.

Company number: 3015516 VAT number: GB 638 5307 26

Current Magazine

Web Edition

Buy
PDF articles

Magazine
Feedback

Podcasts

Competitions

Subscribe

Subscribe
Now

Web Subscription
FAQs

Home

News

Search

New Search

Forum Search

Search Tips

Articles

Reviews

Technique

Sound
Advice

People

Glossary

Help + Support

Forum

Today's
Hot Topics

Forum Channel List

Forum Search

My Forum Home

My Forum
Settings

My
Private Messages

Forum Rules
& Etiquette

SOS TV

Watch exhibition videos,
tutorials,
interviews, masterclasses

Readers
Classifieds

Submit New
Adverts

View My
Adverts

SOS Directory

My
SOS

Change
Password

Change My
Email

Change
My Address

My
Subscription

My
eNewsletters

My
Downloads

Information

About SOS

Contact SOS
Staff

Advertising

Controlled
Circulation

Licensing Enquiries

Magazine
On-sale Dates

SOS Logos
& Graphics

SOS Site
Analytics

Privacy
Policy

All contents copyright © SOS Publications Group
and/or its licensors, 1985-2014. All rights reserved.
The contents of
this article are subject to worldwide copyright protection and reproduction in
whole or part, whether mechanical or electronic, is expressly forbidden without
the prior written consent of the Publishers. Great care has been taken to ensure
accuracy in the preparation of this article but neither Sound On Sound Limited
nor the publishers can be held responsible for its contents. The views expressed
are those of the contributors and not necessarily those of the
publishers.

Web site designed & maintained by PB Associates | SOS | Relative
Media

Home | Tablet
Mag | Podcasts | WIN
Prizes | Subscribe | Advertise | About SOS | Help

Wed 19 Nov 2014

Have an account? Log
in or Register for
free

Email address
Password

Remember me

Stay logged in

Forgot
your password?

Sound On Sound : Est. 1985

Search

New Search

Forum
Search

Search
Tips

News

Articles

Current issue:
November 2014

Recent
issues...

October
2014

September
2014

August
2014

July
2014

June
2014

May
2014

All issues

Reviews

Technique...

Classic
Tracks

Secrets
of the Mix Engineers

Ableton
LIVE techniques

Apple
LOGIC techniques

Cakewalk
SONAR techniques

Cockos
REAPER techniques

Digidesign
PRO TOOLS techniques

MOTU
DIGITAL PERFORMER techniques

Propellerhead
REASON techniques

Steinberg
CUBASE techniques

Sound
Advice

People

Glossary:
Technical Terms

Podcasts

Smart
Guides

Education

Competitions

Forum

Today's
Active Topics

Forum Index

SOS TV

Subscribe

Subscribe
Now

Web Subscriptions
FAQs

Shop

Readers' Adverts

Information

WebExtras