Gitaarversterking en Effecten

Gitaarversterking en Effecten
samengesteld door
Ernst Fliek
In deze reader zijn een aantal artikelen bij elkaar gevoegd over de volgende onderwerpen:
• gitaarversterking
• plaatjes van een aantal bekende versterkers
• effecten uitgelegd
• hoe je effecten aan kan sluiten
• hoe versterkers gebouwd worden
• wat wel en niet waar is over versterkers van toen en nu
• waarom buizen
• een technisch artikel over buizen versus transistoren
Twee schematische voorstellingen moeten de opbouw van de versterker verduidelijken:
• de opbouw van de gitaarversterker
• de effectloop
Aan het eind van het document vind je een aantal internetlinks die je allicht verder op weg helpen.
• internet
Schagen, 08-04-2003
Ernst Fliek.
Guitar amp origins
Bron:
http://www.spectrum.ieee.org/select/0898/tube.html
Guitar amp origins
It was in the '30s that the idea of attaching a transducer, amplifier, and loudspeaker to a guitar first caught
on. Early guitar amps were used primarily with lap guitars, usually made of solid wood or metal with no
resonant body to increase volume. Later amps were adopted by big-band guitarists, and in the '50s,
amplifiers became pretty much mandatory for the amalgam of blues, country music, and jazz known as
rock 'n' roll. Since transistors did not enter wide usage until about 1960, all of the originating styles of rock
guitar were developed on tube amplifiers. Later, musicians discovered that using a gain device before the
guitar amp forced the amp to clip heavily; they liked the resulting sound, and it became the foundation of
hard rock, later called heavy metal.
Since all the past styles of guitar playing are still regarded as musically valid, the market for guitar equipment
has fragmented. Various manufacturers offer arrays of amplifiers, preamps, effects processors, and other
means of electronically processing the guitar signal. Both reproductions of early equipment and innovations
are available. Most remarkably, the basic designs of tube guitar amps tend to be based on a few prototypes
that date from the '50s or early '60s. Extra channels or gain stages are added, tone controls are modified,
sound effects (like reverberation and electronic tremolo) vary, and speaker cabinets become available in
various configurations. Yet the basic circuits keep returning to the same set of paradigms.
No manufacturer of guitar amps has been as influential as Fender. Between 1946 and 1965, founder Leo
Fender and his design team created most of the rock guitar sound in the form of the amplification used
with their solid-body guitars, such as the famous and widely-copied Stratocaster model.
The most primitive design for a Fender amp is the Champ model. Being intended as a low-cost amplifier for
students and beginners, a typical Champ uses a single 6V6GT or 6L6GC power tube. Because of its singleended power stage--and the large amount of second-harmonic distortion thereby engendered--the
Champ had a sound often described as "soft" and "lush." The amp's small, cheap output transformer
saturated easily and gave very poor low-frequency response. Early Champs used 6V6GTs, were extremely
primitive, and had no feedback, while later models had more complex circuits and loop feedback. In spite of
their crudeness, early Champs are now valuable collector's items, and have been much imitated in recent
years. The Champ sound is a standard, typical of many low-cost amps used on early rock 'n' roll records.
The most popular Fender models among serious professionals are the Bandmaster, Twin, Showman, and
Bassman with push-pull 6L6GC or 5881 output tubes. The tone of these models has a ringing quality
much sought after. The peculiar distortions from these amps and their matching speakers, in addition to
the inherently light regulation afforded by tube rectifiers, gives distinctive inherent compression effects.
This combination is what makes possible the infinite sustain effect mentioned earlier.
Even though the Bassman was intended originally for bass guitar, it was widely used for lead guitar and
became possibly the most copied guitar amp in history, especially the 1959 model equipped with four 25cm speaker drivers. The Twin Reverb model is often modified with extra gain stages for more distortion,
producing a fair degree of compression and allowing the lengthy sustain of guitar notes. It served as the
prototype for many modern amps with complex preamp sections.
Starting in 1962, a new sound appeared in Britain. Jim Marshall, a London music dealer, found that
imported Fender amps were popular but too expensive, and so he developed his own. While his first amp
was a copy of the Bassman, he later changed the output tubes to push-pull EL34s. These European tubes
were true pentodes, different in electrical behavior from the beam tetrodes used in Fender amps. With the
new tubes, Marshall's amps took on a tone described as very distorted and "crunchy," which is now
considered the classic British blues-rock sound. Interestingly, the EL34 had reliability problems when
operated in deep clipping for long periods, so in the '70s the U.S. distributor for Marshall amps switched
the output tubes to 6550 beam tetrodes. As the sound of these amps was much more like very powerful
Fenders, some preference arose among U.S. musicians for a "harder" sound than Marshalls give with EL34s.
New distribution in the 1980s had EL34-equipped Marshalls entering the United States, as Jim Marshall
preferred.
The third common guitar-amp design is that of the models AC15 and AC30 made by Vox Amplification
Ltd., London. These were often used in Britain and throughout Europe, most notably by the Beatles at the
peak of their popularity. The AC15 uses two push-pull EL84 output tubes, the AC30 four EL84s. Both
models use self-bias of the output tubes, in Class A operation and with no negative feedback, unlike many
other push-pull guitar amps. The result is a unique tone that varies greatly with string-plucking force.
AC30s were made available with a Top Boost option, adding gain stages for further versatility. The Top
Boost AC30 design is widely imitated by modern amp designers.
Bass guitar amplified otherwise
A bass guitar has different needs from a lead guitar. Bass is used to reinforce the song rhythmically, working
with the melody at a pitch several octaves below. Nearly all bass guitars are solid-body types--simply larger
versions of regular electric guitars with very thick strings. Since the bass sound is not always assisted by
distortion, solid-state designs have come to hold sway over this market. Any amp with a high damping
factor and capable of generating high powers at low frequencies can serve as a bass amp. Yet a ground
swell of interest in tubed bass amplifiers has surfaced since 1990.
Early bass amps, such as Fender models, were essentially little different from guitar amps. Then the Ampeg
SVT was introduced in 1969. It dwarfed previous bass amplifiers, producing 300 W from six 6146 or 6550
tubes. The SVT became a standard much imitated, especially in the last 10 years. Modern tube bass amps
are usually very large, producing 200 W at least from a set of 6550 beam tetrodes--as many as 10 in some
models.
Gitaarversterkers - een paar bekende en oude modellen.
Fender ‘59 Bassman (http://www.fender.com)
Fender ‘65 Twin Reverb Reissue (http://www.fender.com)
Marshall - 1959SLP Super Lead Plexi Head
http://www.1800instruments.com/archive/ghAmps05.htm
Marshall Plexi Superlead
amp from 1968.
by Chris
One of the world truly great pieces of amplifier history and folklore. We road test this 31 year old gal.
This 1959 Super Lead amp (or Plexi as they are affectionately know as due to their screen printed Plexi
glass instrument and back panels) is my baby. It was manufactured in 1968 according to the piece of
paper glued to the chassis and signed by the Marshall workers as they went about checking off their work.
It was an import into Australia and thus came with some EL-34's in the power amp side. On the preamp
side of the equation it comes with 3 12AX7 pre amp tubes, interestingly when you take the back off you
can see a space where the original Valve rectifier tube would have sat before Marshall changed to silicon
rectifiers.
This is the classic Marshall amp with the circuit based on the Tweed Fender Bassman amp, hence it has 4
inputs, no master volume, 3-band EQ and a presence knob and 2 volume knobs. It does not really have
two channels, rather one is a normal input (with high and low inputs just in case the organ player in your
1960's band was going to use it) and the other is a treble boosted channel ala Tweed Bassman. I find that
the old trick of linking the inputs together so that you operate Vol 1 and Vol 2 together gives you just the
right blend of beef and cut when using humbuckers. But experimentation is the key to getting different
sounds from this amp, rather than today's amps that come with pre programmed channels to produce
clean, crunch and lead sound,with this ol' gal you have to use your guitars volume and tone controls and
the different inputs on the amp itself.
Don't be fooled into thinking these amps have a lot of clean headroom all up the gain scale like modern
amps clean channels. Up around 3 on the scale the sound is only semi clean and very Hendrixy in a "Wind
cries Mary" / "All along the watchtower" sort of way. If you want to play louder but keep it clean you have
to juggle amp volume with guitar volume, there's that experimentation I mentioned.
Around 4 on the Volume knobs it starts to dirty up, now you are starting to hear AC/DC coming out the
amp, great for crunch rhythm work. After about 7 the Volume knobs don't really get any louder or dirtier
they just seem to build in weight of sound, now you are into lead territory and Beck, Page, Angus Young
can all be copied. Around this point it will hit a really sweet spot where it will only get "rattier' if you push
the volume knob, it won't get any more distorted or louder.
A lot of people ask me what does this amp sound like? Well the sounds that it produces are like most of
the traditional rock and blues albums you have all heard a million times. Take out you old Cream, Page,
Kossoff LP's and give them a spin, or check out Hendrix LP's for a great example of this amps clean sounds.
Personally this is what this amp does best IMHO, that great semi clean sound, for crunch rhythm just turn
up your guitars volume knobs. For lead sounds step on a distortion box of your choice just like Jimi did. To
get Metal high gain sounds you really need to put a distortion unit inbetween you and your amp. This is
the setup that provides the most versatility from this amp.
People comment on Marshall's (especially the old ones) being unreliable, but I beg to differ. This amp is
built like a brick with good old fashioned wires, and not printed circuit boards, soldering up all the parts.
This makes for some more noise and hum depending on how neatly laid out the wiring is, but you can
always take out the chassis and move the wiring around to reduce hum. This amp has been great for
reliability, but then I have had 3 sets of power tubes in it in the last 6 years, so the local Marshall Tech
here in Adelaide has had it on his bench and checked it over every time the new tubes went in. But still
considering the old girls age, made in 1968 that makes her 31 years old, I would say she's doing great. Lets
see how many mass produced printed circuit board amps will still be deafening people in the year 2030.
Marshall speakerkasten
Links een rechte (straight) kast en rechts een zogenaamde buikkast (angled).
http://wwww.marshallamps.com
Nog een mooie versterker.
Orange Overdrive - http://www.geocities.com/geertjacobs/orange_page.htm
Orange OR120 - http://www.geocities.com/geertjacobs/orange_page.htm
Een erg bekende transistor versterker.
Polytone Mini-Brute III (transistor versterker) de Jazz versterker?
Hoe wordt een versterker opgebouwd? Point to Point of op een printplaat? Wat is waar en wat niet?
PTP Wire Amp Myths You'll Hear
http://www.geofex.com/Article_Folders/pt-to-pt/pt-to-pt.htm
PCB's use traces to connect components that add a small degree of capacitance
between the parts which changes the tone/sound.
It is correct that PCB's use thin copper traces (een betere print gebruikt dikere lagen koper, deze worden
dan ook nog eens vertind) to connect parts and that there is capacitance between all the traces.
However, the implication that point to point wiring does NOT have such parasitic elements is completely
false. Point to point wiring uses "wires" to connect components; these "wires" also add a small degree of
capacitance between the parts. In fact, for some instances, hookup wire can have MORE wire-to-wire
capacitance per unit length than PCB traces. There are no wires that *don't* have self inductance,
capacitance, resistance and all the other ills that every electronic part is heir to. Which has more? It
depends... you have to know the mechanical specification of exactly where every part is in relation to every
wire and the chassis to compute the capacitance for either traces or wires. A well designed PCB with
attention paid to where the traces go may have bigger or smaller capacitances to other traces and parts.
As to this necessarily degrading tone - almost always no. The capacitances may or may not have any
audible effect, and if there is an effect on the tone, it may or may not be positive, or may or may not be
swamped out by the actual parts instead of the wires/traces. The only way to tell is to figure out what the
capacitances actually are and see if it makes a difference - what's the capacitance of a signal wire lying
against the chassis for 6"? Is it more or less than the capacitance of a PCB trace for 6"? Hmmm... can't tell
without measuring and seeing what this capacitance couples together in the amp.
Companies used to use carbon comp resistors which yield a richer, more complex tone.
Ah, yes. The old "They don't make them like they used to". They really don't make them the same any
more - thank God. Resistors used to be 20% tolerance, drifted another 5-10%, and had a "hole in the
middle" where all the ones that did by accident get within 5% and 10% of the nominal value had been
selected out. A 20% resistor was guaranteed to not be any closer that 11% to the nominal value because
all the 10% and better ones had been selected out by the manufacturer. Old carbon comp processes had
build in distortions, rectification effects, and so on. Process control is much better than it used to be. The
differences in carbon comp and modern film resistors are real, but that whole thing is a topic for another
article. Suffice it to say that just because it's carbon comp, it's not tone magic. Carbon comps were the
cheapest available resistors in the "golden age", and were used for that reason. Leo Fender was a
businessman, not a guitarist.
The quality of the parts has been reduced / Companies used to use better quality caps.
Another one. This one is pure nonsense. Modern capacitors are more stable, more consistent, have closer
tolerances, lower drift with time and soldering, and so on.
Ah, but the old paper capacitors were tone magic/mojo magnets/whatever.
Nonsense. Old paper capacitors had higher ESR and ESL than some modern capacitors, lower than others.
It's entirely possible to get *better* modern caps and then add resistors to "ugly them down" to what the
older caps did. You do have to know what you're doing, though.
Modern parts that cost next to nothing when bought bulk quantities are loaded by
computers.
Leo Fender would have jumped on that!! He'd have loved it. As it was, he used the cheapest parts and
labor that would produce his amps reliably enough to sell well. Leo was a businessman, not a guitarist.
Cheaper parts and pcb's lead to an amp or effect that sounds stiff, lifeless or one
dimensional compared to the earlier ones
Huh? Why? What parts affect the tone? This is a form of creeping superstition. Most audio research
indicates that the terms "stiff", "lifeless", "tiring", and so on are associated with low levels of intermodulation
distortion. This is a circuits thing, not a wiring thing, no matter how it's done.
Handwired, point to point will always sound better than any stock PCB amp.
Lets face it, in the 50's and early 60's amps were wired point-to-point, because it was cheap...the skilled
labor required to solder wires to the turrets was cheaper than buying PCB making equipment, which was
still experimental at the time. But there's no real evidence that you can't do as well or better with a PCB.
And the PCB's will come out the same way, every time. No little lead dress problems making some of your
amplifiers oscillate when the rest of them are OK.
‘Point to Point ‘ wiring.
PCB of Gedrukte Bedrading
Opbouw van de gitaarversterker
De versterker
VERSTERKER
De twee onderdelen van de versterker
VOOR
VERSTERKER
pre amp
EIND
VERSTERKER
power amp
Versterker met send en return
pre amp
p ow er amp
VOOR
VERSTERKER
SEND
EIND
VERSTERKER
RETURN
Seriële en parallelle effectloops
Seriële effectloop
Bij de seriële loop, gaat het gitaarsignaal via de send de versterker uit, gaat naar
een effectapparaat, wordt daar bewerkt en komt via de return de versterke r
EIND
VERSTERKER
VOOR
VERSTERKER
SEND
RETURN
EFFECT
IN
OUT
Parallelle effectloop
Bij de parallelle effectloop blijft het originele signaal in de versterker, wordt er
signaal van afgetakt en naar de send gestuurd. Na bewerkt te zijn gaat dit signaal
terug naar de return en wordt weer bij het origineel gemengd.
+
direct signaal
VOOR
VERSTERKER
SEND
RETURN
EFFECT
IN
OUT
EIND
VERSTERKER
Korte beschrijving van gitaar effecten
Bron;
http://www.geofex.com
AMPLITUDE BASED EFFECTS
Volume control
Manual level control. Twist the knob, the sound gets louder or softer. Examples : Morley and DeArmond
Volume pedals
Tremolo
Cyclical variation of volume by a low frequency oscillator of some sort; parameters are waveform of the
LFO, LFO frequency, and depth of modulation; note that while the terms tremolo and vibrato are often
used interchangeably, tremolo is actually variation in loudness, vibrato is variation in pitch or frequency.
Examples : Diaz tremodillo (new), Boss PN-2 Pan tremolo
Auto tremolo
tremolo where the modulation frequency is varied by some feature of the input signal, generally amplitude.
Panning/ping-pong
generalization of tremolo to more than one channel; as one channel goes down in level, another goes up.
With non-square LFO waveforms, gives the effect of the sound source moving from place to place in
stereo or more channel setups. Examples : Boss PN-2 Pan tremolo (1989), Ibanez Flying Pan (1976?)
Gating/repeat percussion
tremolo with 100% modulation of the signal by a square wave. With exponentially decaying waveforms
(guitar is a good one), gives the effect of striking the same note again at decreasing levels. Some Thomas
Organ Vox amps have this as a built in effect. Examples : Vox Repeat percussion ('70), Walco "sound goround" ('75?
Compression
makes soft inputs louder, and loud ones softer, giving a one-level kind of sound with lessened dymanics.
This is effectively volume control with the level determined by the negation of the averaged envelope of
the input level. Early compressors were often called "Sustain" pedals. The Electro-Harmonix Black Finger
was among the first compressors in the early 1970s, along with the Maestro sustainer. Examples : MXR
Dynacomp ('74 - present), Boss CS-1 to 3 ('78 - present)
Expansion
Makes loud sounds louder and soft ones softer. Effectively volume control with the level determined by the
averaged envelope of the input level. Compression and expansion can be complementary, as in >>
com(pression/ex)panding for noise reduction. Asymmetric compression/peak compression
Only the peaks of the input waveforms get compressed, not the overall level of the waveform envelope.
Effectively, there is no averaging of the envelope and the instantaneous waveform level is compressed. This
amounts to a much softer form of clipping, and is part of the tube sound, since tube with a soft B+ supply
are prone to this.
Noise gating
modulates the output off when the input level is below a threshold. The modulation may be a square
wave, or a variation of expansion where the low level inputs are "expanded" down into silence, which gives
a less abrupt transition. Examples : Electro Harmonix silencer, MXR noise gate line driver ('70s)
Attack delay
A variation of noise gating where the transition to "on" from the "off" or no signal state is slowed. This
gives an output which perceptibly rises in level with each new note envelope, reminiscent of a tape
recording played backwards. Examples : Electro Harmonix Attack delay ('79), Boss SG-1 Slow gear ('80)
ADSR
Term borrowed from the synthesizer folks; stands for Attack Decay Sustain Release, which is the most
general way to describe a musical envelope. It is possible to generate an artificial ADSR envelope for a
musical note to help fool the ear as to which instrument generated the note. Example : Electro Harmonix
Micro Synth (sort of) ('82)
Electro Harmonix - Micro Synthesizer
Limiting
Like compression, but operates on signals over some threshold only. Well suited to keep an input from
going over some level, but un-processed below that level, as in getting signals on tape without overloading
the tape. Example : MXR Limiter ('80)
Auto swell
generally, a rise in level from some starting level to a final level when keyed manually or electronically. Can
effectively add sustain to some notes and not others when keyed manually, or can add a "swell" in volume
over a run of notes, or can help with presetting the level of a lead.
WAVEFORM DISTORTION EFFECTS
Ibanez TS9 Tube Screamer, een beroemd pedaaltje
Symmetrical clipping
For a given input waveform, say a sine wave, the tops and bottoms of the waveform are clipped equally,
symmetrically. Although the musical implications are more involved than this simplistic explanation, for a
simple sine wave, symmetrical clipping generates only odd-order harmonics, giving a reedy, or raspy sound
to the resultant waveform.
The hardness or softness of the clipping matters. Hard clipping results when the output wave equals the
input up/down to a certain level, then stays at the clipping level until the input drops below the clipping
level again, giving perfectly flat tops and bottoms to the clipped output. Soft clipping has no abrupt
clipping level, but gently rounds the top/bottom of the output wave so the waveform is "softly" rounded
on top/bottom, not flat-topped. Some solid state devices actually flat top, then invert, producing a hollow
topped output waveform at hard clipping. There is a continuum of clipping hardness, depending on the
circuitry used to clip. Soft clipping emphasizes the lower- order harmonics, the third and fifth, etc. Hard
clipping has a mix slewed to the higher order seventh and up harmonics, which are harsher sounding.
Intermodulation distortion, the production of sum and difference frequencies from frequencies in the
input waveform, varies with the amount and hardness of clipping. Intermodulation sounds harsh and ugly.
The amount of intermodulation is a characteristic of the circuit that produces the distortion.
Electro Harmonix - Big Muff π
Asymmetrical clipping
The top(or bottom) of the waveform is clipped more than the bottom (top) half. This causes the
generation of both even and odd harmonics, in contrast to symmetrical clipping's odd-order only. The
even harmonics are smoother and more musical sounding, not as harsh as the odd ones. The hardness of
the clipping and the degree of asymmetry affect the sound. The more asymmetrical, the more pronounced
the even-order harmonics; the harsher the clipping, the more the harmonics are slewed toward higher
order. See Half Wave Rectification.
Tubes in general produce asymmetrical distortion unless the circuitry is set up to remove them, as happens
in push-pull.
Infinite limiting
In essence, the waveform is amplified "infinitely" and hard and symmetrically clipped, producing a
rectangular output waveform which shares only the zero crossings with the input waveform. Sounds
buzzy and synthesizer-ish.
Examples : Mosrite Fuzz-rite ('68), Early Maestro Fuzz tones ('64-'70s)
Half wave rectification
Half wave rectification represents the logical conclusion of asymmetrical clipping. One half of the waveform
is flat, the other half is unchanged. This produces a prominent second harmonic, heard as an octave. There
are analog devices which produce an octave effect injust this way; I have heard that the "Octavia" effect is
based on this.
Examples : Foxx tone machine, Tycobrahe and R.M. octavias, Super Fuzz. ('70s)
Full wave rectification
In full wave rectification, one half of the input waveform is "folded" to the opposite polarity, producing an
output with a net DC component, none of the original fundamental frequency of the input waveform, and
only the second and higher harmonics of the original input frequency. Produces very strong octave of the
input waveform, as well as a slew of even-, odd-, and intermod- distortion products when more than a
single frequency is the input ( as is the case for all musical instruments). Arbitrary waveform generation
This effect generates a completely new waveform of arbitrary shape which shares the same frequency as
the input waveform. Guitar synthesizers do a version of this.
FILTER/FREQUENCY RESPONSE EFFECTSEQ
Tone controls
Allow you to cut or boost the highs, lows, mids etc. Tend to be broad-brush kinds of controls - all the
"high's" get raised or cut. Range is typically +/- 12 to 20 db boost/cut.
Example : the tone control on your guitar.
Treble/mid/bass boost
Like an additional eq control, but tends to be narrower in frequency range, and perhaps more boost
range, no cut. Electro Harmonix made a range of boosters starting in the early 70's to boost treble
(screaming tree), Bass (Mole) or everything (linear power booster). Very simple, similar circuits were used.
Examples : Dan Armstrong purple peaker, Vox treble booster. ('72)
Cabinet simulation
A filter network designed to mimic the two- or four-pole low frequency rolloff of a guitar speaker cabinet,
usually to get that "miked cabinet" sound into a PA without really miking a cabinet.
Examples : Tech 21 Sans Amp (new)
Resonator
A filter with a boost in frequency at a narrow range of frequencies. This sounds like a wah pedal when the
pedal is not being moved.
Example : Boss SP-1 Spectrum ('79)
Wah
A resonator that can have its center frequency moved up or down in frequency by moving a pedal. The
"wah" name comes from the way it mimics the moving resonance of the human vocal tract in speech as the
sound "wah" is made. The first wah wah was the Vox Clyde McCoy, which was originally designed to
emulate the sound of trumpet players using a mute(!).
Example : Crybaby ('60s to present)
Auto wah or "Envelope Follower"
A wah filter where the center frequency is determined by the loudness of the input signal, making a moving
resonance on every note. The Mutron-III was about the first envelope effect. It was made in the mid
1970s, and is still thought to be the best ever made by many players.
Example : Electro Harmonix Doctor Q ('76), MXR envelope filter ('76)
Tremolo-wah
Wah where the center frequency is moved back and forth cyclically, as though the pedal was connected
to a motor or some such. This can generate effects similar to a rotating speaker or phasing.
Example : Boss Auto Wah AW-2 (80s to present)
"vibrato"
A cyclical variation in the basic frequency of the input signal, similar to the effect of moving the whammy
bar on a suitably equipped guitar. True vibrato as an add on effect requires some kind of time delay, and
was hard to do until analog (and now digital) delays came to be. The Uni-vibe, (trictly speaking a phase
shifter) dsigned in about 1967, had a vibrato setting. However it is more famous fo it's "chorus" setting.
Examples : Boss CE-1 Chorus Ensemble ('77)
Phase shifting
This effect is a filter response generated by using long phase delays and mixing with the original signal to
cause a number of deep notches and/or peaks in the overal filter response. This mimics the larger number
of notches and peaks caused by true time delayed flanging. Most simple phase shifters or phasers do this
by generating two notches, although some pedals make four notches. Flangers may make many notches.
Phasers may also incorporate feedback to sharpen up the effect of the notches.
About the first phaser was the Maestro PS-1 designed by Oberheim in about 1970. The MXR Phase-90
and very inexpensive but good sounding Electro Harmonix "Small Stone" were used on too many pop
songs in the 1970s.
Examples : Boss PH-2 Super phaser (new), MXR phase-45 90 and 100 ('70s)
TIME DELAY EFFECTS
Echo/Delay
Simply put, a delay takes an audio signal, and plays it back after the delay time. The delay time can range
from several milliseconds to several seconds.
Analog (MXR Analog Delay, EH Memory Man) or Digital delays (Boss DD-3)
Reverb
Reverberation is the result of the many reflections of a sound that occur in a room. From any sound
source, say a speaker of your stereo, there is a direct path that the sounds covers to reach our ears. But
that's not the only way the sound can reach us. Sound waves can also take a slightly longer path by
reflecting off a wall or the ceiling, before arriving at your ears, as shown in Figure 1. A reflected sound wave
like this will arrive a little later than the direct sound, since it travels a longer distance, and is generally a
little weaker, as the walls and other surfaces in the room will absorb some of the sound energy. Of course,
these reflected waves can again bounce off another wall before arriving at your ears, and so on. This series
of delayed and attenuated sound waves is what we call reverb, and this is what creates the 'spaciousness'
of a room.
It's very tempting to say that reverb a series of echoes, but this isn't quite correct. 'Echo' generally implies a
distinct, delayed version of a sound, as you would hear with a delay more than one or two-tenths of a
second. With reverb, each delayed sound wave arrives in such a short period of time that we do not
perceive each reflection as a copy of the original sound. Even though we can't discern every reflection, we
still hear the effect that the entire series of reflections has.
So far, it sounds like a simple delay device with feedback might produce reverberation. Although a delay
can add a similar effect, there is one very important feature that a simple delay unit will not produce - the
rate of arriving reflections changes over time, whereas the delay can only simulate reflections with a fixed
time interval between them. In reverb, for a short period after the direct sound, there is generally a set of
well defined and directional reflections that are directly related to the shape and size of the room, as well
as the position of the source and listener in the room. These are the early reflections (also called the 'early
echoes' despite the general meaning of the word 'echo'). After the early reflections, the rate of the arriving
reflections increases greatly. These reflections are more random and difficult to relate to the physical
characteristics of the room. This is called the diffuse reverberation, or the late reflections. It is believed that
the diffuse reverberation is the primary factor establishing a room's 'size', and it decays exponentially in
good concert halls. A simple delay with feedback will only simulate reflections with a fixed time interval
between reflections. An example impulse response for a room is depicted in Figure 2. (For those who are
not sure what an impulse response is, think of it like this. If you consider a small piece of a sound, each
vertical line marks when that same piece of sound is heard again, and the height of the columns is how
loud the sound is at that time.)
Spring reverebs (Fender) or Digital/Analog reverb pedals (Boss RV-3)
True vibrato
Boss VB-2 ('80s), Electro Harmonix clone Theory ('78)
Flanging
Flanging is created by mixing a signal with a slightly delayed copy of itself, where the length of the delay is
constantly changing. This isn't difficult to produce with standard audio equipment, and it is believed that
flanging was actually "discovered" by accident. Legend says it originated while the Beatles were producing
an album. A tape machine was being used for a delay and someone touched the rim of a tape reel,
changing the pitch. With some more tinkering and mixing of signals, that characteristic flanging sound was
created. The rim of the reel is also known as the 'flange', hence the name 'flanging'.
Most modern day flangers let you shape the sound by allowing you to control how much of the delayed
signal is added to the original, which is usually referred to as a 'depth' control (or 'mix'). Figure 1 is a
diagram of a simple flanger with this depth control.
When we listen to a flanged signal, we don't hear an echo because the delay is so short. In a flanger, the
typical delay times are from 1 to 10 milliseconds (the human ear will perceive an echo if the delay is more
than 50-70 milliseconds or so). Instead of creating an echo, the delay has a filtering effect on the signal,
and this effect creates a series of notches in the frequency response, as shown in Figure 2. Points at which
the frequency response goes to zero means that sounds of that frequency are eliminated, while other
frequencies are passed with some amplitude change. This frequency response is sometimes called a comb
filter, as its notches resemble the teeth on a comb. (see derivation of frequency response)
http://www.harmony-central.com/Effects/Articles/Flanging/
A/DA flanger ('76), MXR Flanger ('78), EH Electric Mistress ('76)
Chorus/ADT
Just as a chorus is a group of singers, the chorus effect can make a single instrument sound like there are
actually several instruments being played. It adds some thickness to the sound, and is often described as
'lush' or 'rich'.
http://www.harmony-central.com/Effects/Articles/Chorus/
Boss Chorus Ensemble CE-1 ('75), MXR stereo Chorus ('78)
Slapback
EH full double tracking ('80), any delay pedal set to short delay time.
Reverse echo/reverb
Most digital effects processors
Sampling
Boss DD-3 (new), DOD PDS-8000 (8 secs sampling, '89) and DFX91 (1 sec, new)
OTHER MISCELLANEOUS EFFECTS
Octave division
Takes the fundamental frequency of an input signal, divides it by two, and creates an octave-lower,
sometimes a two-octave lower signal, which are usually mixed back with the original signal. This is most
often done with digital logic flipflops to divide the signal by two/four after squaring up the input to drive
the flipflops. This provides outputs that are substantially square waves, sounds like fuzz bass.
Some kind of filtering is usually provided to tame the sharp buzz of the square waves. The simple dividers
like this get very confused when fed more than one tone at once, so single note runs are all that is really
practical - unless you like confused effects.
About the first lower octaver was the Maestro OB-1 octaver, which tracked quite poorly, but the note
was not too fuzzy. The mid 1970s Electro Harmonix "octave multiplexor" (1 octave down) and MXR
"Blue box" (2 octaves down) were more successful but the lower octave is pure fuzz.
Harmony generation
Generation of other notes at musically-interesting intervals along with your notes. The classic device to do
this is the Eventide Harmonizer. It is very difficult to do this electronically so that the effect produces
musically-useful sounds consistently, hence Eventide's high price.
There is a new (1996) Boss pedal harmonizer now available which allows selecting the key in which you
are playing, and calculates and plays the desired harmonic (3rd, 5th, etc). It is not cheap but several times
cheaper than an Eventide.
Phase lock tracking
An electronic circuit called a "phase locked loop" can produce an output signal that is exactly an integer
multiple or small-numbers fractions of a reference signal in frequency. You can generate: a signal that
follows your notes, perhaps lagging a little with a glide onto the note an octave or two above a
third/fifth/seventh, etc. above or below your notes.
Sounds kind of like a computer playing harmony with you.
The outputs are usually square wave or filtered square wave, and sound kind of synthesizer-y. Modern alldigital MIDI-fied effects do something like this in their computer processors, and may not be as limited in
output waveform.
Noise addition
Noise (hiss, rumble, etc) is deliberately added to the input signal. If this is done with restraint and matching
the input signal envelope, it can add a breathing effect like the hiss of air in a flute.
Filtered low frequencies can add a growling quality.
Talk box
This effect is produced by using a small amp to produce sound that is conducted into your mouth by a
tube, so you can mouth the words to a song, using your vocal tract resonances to shape the instrument
sound, which is then picked up by a microphone. This is the archetypical "talking guitar".
One of the first talk boxes was "The Bag" by Kustom Electronics (not the amp company). This came in a
psychodelic upholstered wine flask shaped bag, and was used by Jeff Beck. Peter Frampton popularized the
talk box in the mid 1970s with "Do you feel like I do".
Examples : Heil sound talk box (new), Electro Harmonix Golden Throat ('70s)
Voice tracking (vocoder)
Ring modulation (Double Side Band Suppressed Carrier generation) Single Side Band Suppressed Carrier
generation
Example : Roland vocoder
COMMON COMBINATIONS
Leslie
Vibrato, tremolo, varying filtering generated by rotating speaker. There are also several "leslie simulator"
effects such as the new Korg unit, and 1970's Multivox "Full Rotor" and "Little David".
Example : Maestro Rover ('75), RMI Roto Phaser ('76)
http://theatreorgans.com/hammond/faq/mystery/mystery.html
Aphex/Exciter
filtering, selective frequency band distortion Enhancers
Split the signal into a few bands, slightly distort some, remix.
Effects Order (in welke volgorde sluit je je effecten aan?)
This is a perennial question on all guitar oriented forums - what order do I put my effects in? While there
are some simple guidelines, there is no "right" way to do it. It's all a matter of taste and your personal tone.
Let your ears be the final arbiter.
The order of effects that produce the sounds most people have become accustomed to hearing is this:
* Amplitude altering effects
* Pre-distortion EQ
* Distortion(s)
* Post Distortion EQ
* Other tone controls
* Small time delays and Phasers
* Longer time delays - chorus
* Reverb and echo, tape delays, etc.
There is a rationale for the placement of each effect in this order; it goes like this:
Amplitude altering effects
As simple as your guitar's volume knob, or as complicated as fancy compressors, attack-delay or other
note-shaping device. The idea here is that the basic "shape" of the note that will interact with the later
distortion devices gets set for the best tone at that level. Because distortions are level sensitive, the higher
the level that comes out of an amplitude device, the more it will be distorted in any following distortion
devices, and vice versa. A distortion following a level changing device converts the level-altering device into
a distortion-intensity modulator - and that reverts to level changes if you switch the distortion out.
Pre-distortion EQ
Once again, as simple as your guitar's tone control (which is really a simple treble-cut filter) or as
complicated as a parametric EQ; pre-distortion EQ sets up which frequencies are loudest - and the louder
the frequency, the more that a following distortion will affect it. As I mentioned before, distortions are level
sensitive devices - anything under the level at which distortion starts will be largely unaffected. Anything
over the threshold will be distorted. So by boosting things we want distorted and NOT boosting things
we don't want distorted, we can select the things that get distorted and have a much more animated
sounding distortion.One of the most recognizable uses of this was Jimi Hendrix' use of a wah pedal (which
is really a sweepable resonant filter - see the Technology of the Wah Pedal at GEO) before a Fuzz Face. A
wah boosts one band of frequencies a lot, and if the levels are set right, the frequencies in the boosted
range will be distorted most.
Distortion(s)
The Ronco Veg-a-Matic of the sound world, distortions take whatever signal is coming in and slice it into
analog coleslaw. In doing this, they add harmonics and intermodulation products that were not present in
the original signal. This usually results in a hotter high end, as it adds more signal bits at higher frequencies
that were originally present.
Post Distortion EQ and tone controls
Once the distortion has had its way with the signal and inserted a hash of harmonics into it, post
distortion EQ can step in and select which bits out of this sonic stew get heard. As in many things musical,
this started out unnoticed, just the nature of the beast. A 10" or 12" speaker in a cabinet has a frequency
rolloff that starts between 4kHz and 6kHz, and is quite steep. This puts a serious cut on any real high
frequency content from guitar. In fact, many "speaker simulators" are just multipole lowpass filters with
turnover frequencies in the 4K to 6K range, and do a creditable job.Having noticed the post-distortion
tone effect, we can mess with it deliberately, of course. Distortion devices make for lots of high frequency
harmonics. We can cut, boost, trim, notch, and otherwise shape what the distortion device turns
out.Notice that Pre-Distortion EQ changes what gets distorted in the first place. Post distortion EQ can
only cut and trim on what has already been created in the distortion device. You should try it both ways or both ways at once!. Notice that Post distortion wah sounds very different from pre-distortion wah. Try
it! Anything else that does frequency shaping goes in here as well - remember the interaction of level
boost-cuts with distortion.
Small time delays and Phasers
These add a spacious sound by causing multiple notches in the signal at specific frequencies. The ear is
fooled into thinking it's in an acoustic space that has odd cancellations and echoes.Longer time delays chorus
Reverb and echo.
Some combinations and the rationales behind them:
Compressor before distortion
Gives a "smoother" distortion sound because the signal level the distortion gets has less variation - the
compressor wipes off more of the signal changes, so the distortion works mostly at one level, and the tone
quality of the distortion changes less as the note decays. The disadvantage is that the hiss of the
compressor is further amplified by the distortion, so this setup is noisier than either by itself.Distortion
before compressor
The compressor adds little but hiss, because the distortion already sets up a fairly fixed output level. The
tone quality changes as the distortion would without the compressor.Distortion before time delay
The subtleties of the time delay, chorus, flange, etc. are generated after the distortion's harmonic hash, so
the nuances of the delay can be heard.Time delay before distortion
The distortion's harmonic generation tends to fill in the response notches the time delay created, usually
less acceptable.
In the end, only your ears can determine what your sound needs to be. Experiment! Find *your* sound. In
the end, the only right way is your way.
Why Vacuum Tubes?
Bron:
http://www.dwfearn.com/whytubes.htm
by Douglas W. Fearn
For some of us in audio, tubes never really went away. We go back to the days when all equipment was
tube equipment. (My first job, in the late 1960s, was at a Philadelphia radio station that was 100% "hollow
state" from microphone to antenna.) I admit that I was thrilled when the first solid-state tape machines
and recording consoles appeared, even though they didn't sound as good as the old tube gear they
replaced. The features were dazzling, so many of us were willing to tolerate a grungier sound, or pretend
we didn't hear it -- at least for a while. (Sounds sort of like the digital vs. analog controversy now raging in
some circles, doesn't it?)
It seems that many of us who used tube gear in the past, and a growing segment of younger engineers and
producers, have (re-)discovered tubes. Do they really sound better, or at least different? And if so, why?
Much (and maybe all) of the answer is contained in a paper in the May 1973 Journal of the Audio
Engineering Society by Russell O. Hamm, Tubes Versus Transistors -- Is There an Audible Difference? The
quotes in this piece are from that article. Hamm found that the output level of studio microphones under
typical recording conditions contains peaks far in excess of what VU meters display. Everyone knows that,
but the peaks, as measured with an oscilloscope, are really high, easily exceeding 1 volt! The tube or
transistor used in a condenser microphone, or in a microphone preamplifier, often will be driven into
severe overload by these peaks. The peaks are short, so the sound isn't grossly distorted-sounding; but the
distorted peaks do affect what we hear. All preamps (and condenser mic electronics) are overloaded by
these peaks, but tubes handle it differently than solid-state devices.
When transistors overload (in a discrete circuit or in an op amp), the dominant distortion product is the
third harmonic. The third harmonic "produces a sound many musicians refer to as 'blanketed.' Instead of
making the tone fuller, a strong third actually makes the tone softer." On the other hand, with tubes
(particularly triodes) the dominant distortion product is the second harmonic: "Musically the second is an
octave above the fundamental and is almost inaudible, yet it adds body to the sound, making it fuller."
There's a lot more revealing information in Russ Hamm's paper, and if you want more details about why
tubes often sound better, it's worth finding a copy.
Another difference between tubes and solid-state devices (including FETs) is the load they provide to the
source. This is particularly significant with condenser microphone capsules. Even though an FET has an
input impedance similar to tubes (in the megohms), for some reason condenser elements just sound better
going into a tube. Is it input capacitance? Or some sort of dynamic loading factor?
If you ask enough people (particularly audiophiles), you'll start to discover some truly metaphysical theories
on why tubes sound better. Some explanations utilize quantum theory, some are simply wrong, while
others are just plain bizarre. But in my own circuit designs I've encountered some odd phenomena that I
can't explain. For example, in a mic preamp prototype, I discovered that performance measurements were
different if the unit was upside down. After I tried to eliminate all possible environmental factors, the
difference persisted, even with a variety of tubes, and in different buildings. It was audible, too, though
subtle. Was it caused by gravity slightly shifting the internal tube elements? The earth's magnetic field? I still
don't know.
Tubes sound better because their distortion products are more musical. Tubes provide a more
appropriate load to transducers. Those are the fundamental reasons why tubes sound better, but is there
more?
Tubes Versus Transistors - Is There an Audible Difference?
by Russell O. Hamm
Sear Sound Studios, New York, N.Y.
Presented September 14, 1972, at the 43rd Convention of the Audio Engineering Society, New York.
Engineers and musicians have long debated the question of tube sound versus transistor
sound. Previous attempts to measure this difference have always assumed linear
operation of the test amplifier. This conventional method of frequency response,
distortion, and noise measurement has shown that no significant difference exists. This
paper, however, points out that amplifiers are often severely overloaded by signal
transients (THD 30%). Under this condition there is a major difference in the harmonic
distortion components of the amplified signal, with tubes, transistors, and operational
amplifiers separating into distinct groups.
INTRODUCTION:
As a recording engineer we become directly involved with the tube sound versus transistor sound
controversy as it related to pop recording. The difference became markedly noticeable as more
solid-state consoles made their appearance. Of course there are so many sound problems related to
studio acoustics that electronic problems are generally considered the least of one's worries. After
acoustically rebuilding several studios, however, we began to question just how much of a role
acoustics played.
During one session in a studio notorious for bad sound we plugged the microphones into Ampex
portable mixers instead of the regular console. The change in sound quality was nothing short of
incredible. All the acoustic changes we had made in that studio never had brought about the vast
improvement in the sound that a single change in electronics had. Over a period of several years we
continued this rather informal investigation of the electronic sound problem. In the past, we have
heard many widely varied theories that explain the problem, but no one, however, could actually
measure it in meaningful terms.
PSYCHOACOUSTICS
Anyone who listens to phonograph records closely can tell that tubes sound different from
transistors. Defining what this difference is, however, is a complex psychoacoustical problem. Any
investigation of this admittedly subtle phenomenon must really begin with a few human
observations. Some people try to point out and describe valid differences. Others just object to the
entire thesis and resort to spouting opinions. It is the listener's job to sort out the facts from the
fiction.
Electrical engineers, especially the ones who design recording equipment, can prove that these is no
difference in tube or transistor sound. They do this by showing the latest specification sheets and
quoting electronic figures which are visually quite impressive. It is true, according to the parameters
being measured, that these is only a marginal difference in the signal quality. But are there some
important parameters which are not being measured? One engineer who admits that there might be
some marginal difference in the sound, says, "You just have to get used to the nice clean sound of
transistors. What you've been listening to on tubes is a lot of distortion." Of course the question
which comes to mind is, What is this distortion and how is it measured?
Psychoacoustically, musicians make more objective subjects than engineers. While their terms may
not be expressed in standard units, the musician's "by ear" measuring technique seems quite valid.
Consider the possibility that the ear's response may be quite different than an oscilloscope's.
"Tube records have more bass. . . . The bass actually sounds an octave lower," says one rock
guitarist. A couple of professional studio players have pointed out on numerous occasions that the
middle range of tube recordings is very clear, each instrument has presence, even at very low
playback levels. Transistor recordings tend to emphasize the sibilants and cymbals, especially at low
levels. "Transistor recordings are very clean but they lack the 'air' of a good tube recording." "With
tubes there is a space between the instruments even when they play loud . . . transistors make a lot of
buzzing." Two people commented that transistors added a lot of musically unrelated harmonics or
white noise, especially on attack transients. This same phenomenon was expressed by another
person as a "shattered glass" sound that restricted the dynamics. It was generally agreed that tubes
did not have this problem because they overloaded gently. Finally, according to one record producer,
"Transistor records sound restricted like they're under a blanket. Tube records jump out of the
speaker at you. . . . Transistors have highs and lows but there is no punch to the sound."
When we heard an unusually loud and clear popular-music studio recording, we tried to trace its
origin. In almost every case we found that the recording console had vacuum- tube preamplifiers.
We are specific in mentioning preamplifiers because in many cases we found hybrid systems.
Typically this is a three- or four-track console that is modified with solid-state line amplifiers to feed
a solid-state eight- or sixteen-track tape machine.
Our extensive checking has indicated only two areas where vacuum-tube circuitry makes a definite
audible difference in the sound quality: microphone preamplifiers and power amplifiers driving
speakers or disc cutters. Both are applications where there is a mechanical-electrical interface.
As the preliminary basis for our further investigation we decided to look into microphones and
preamplifier signal levels under actual studio operating conditions. Hoping to find some clues here
we would then try to carry this work further and relate electrical operating conditions to acoustically
subjective sound colorations. Our search through published literature showed that little work has
been undertaken in this area. Most microphone manufacturers publish extensive data on output
levels under standard test conditions [1], but this is rather hard to convert to terms of microphone
distances and playing volumes. Preamplifier circuit design is well covered for noise considerations
[2], but not from the standpoint of actual microphone operating levels. Distortion has been treated in
numerous ways [3-5], but with very few references to musical sound quality [10].
MICROPHONE OUTPUT LEVELS
To get a rough idea of the voltage output from different types of microphones, an oscilloscope was
paralleled across inputs of a console. During the normal popular-music type sessions, peak readings
of 1 volt or more were common, especially from closeup microphones on voice and drums. Due to
the linear voltage scale, oscilloscope measrements over more than a 10-dB range are difficult. By
building a simple bipolar logarithmic amplifier, the useful measuring range was extended to about
four decades (Fig. 1). Considerable studio observation finally led to the construction of a peak
holding type decibel meter. This circuit retained transient peaks of more than 50 microseconds within
2-dB accuracy for about 10 seconds; long enough to write them down. Using the logarithmic
oscillo- scope display and the peak meter together proved very useful in gathering a wealth of data
about real-life microphone signals.
Fig. 1. Simplified bipolar logarithmic amplifier schematic.
Table I shows the normal peak outputs from several popular types of studio microphones. All the
readings are taken with the microphone operating into the primary of an unloaded transformer.
Pickup distances are indicated for each instru- ment and were determined by normal studio practice.
Table II is an abridgement of a similar studio done by Fine Recording, Inc., several years ago.
Details of this test setup are not available but the readings are probably taken without the 6-dB pad
commonly used on the U-47 microphone today. Some calculations based on the manufacturer's
published sensitivity for these microphones indicated that acoustic sound-pressure levels in excess
of 130 dB are common. While the latest console preamplifiers have less noise, less distortion, and
more knobs than ever before, they are not designed to handle this kind of input level. In most
commercially available preamplifiers, head room runs on the order of +20 dBm, 1 and gain is
commonly set at 40 dB. With these basic parameters it is clear from the data shown in Tables I and II
that severe overloads can occur on peaks from almost all instruments. For example, a U-87
microphone gives a peak output of -1 dBm from a large floor tom. Amplification by 40 dB in the
microphone preamplifier results in an output swing of +39 dBm, or almost 20 dB above the
overload point. Logically a peak of this magnitude should be severely distorted.
Table 1. Peak microphone output levels for percussive sounds. Microphone Voltage, Open Circuit,
dB Ref. 0.775 V
Instrument
Distance (in.) U-87 U-47 77-DX C-28 666
Bass drum (single head)
6
0
-6
-9
-15
-1
Large tom tom
12
-1
-6
-9
-10
-5
Small tom tom
12
-1
-5
-7
-9
-1
Piano (single note)
6
-25
-29
-38
-35
-32
Piano (chord)
6
-23
-27
-36
-33
-33
Orchestra bells
18
-16
-25
-33
-33
-30
Cow bell
12
-10
-12
-29
-19
-15
Loud yell
4
0
-11
---
-10
-10
* U-87 and U-47 by Neumann, 77DX by RCA, C-28 by AKG, 666 by Electro-Voice.
Most recording consoles today have variable resistive pads on the microphone inputs to attenuate
signal levels which are beyond the capabilities of the preamplifier. The common use of these input
pads supposedly came about with the advent of loud rock music; however, this is not true in fact.
For some 20 years it has been common to use a Neumann U-47 microphone for close microphone
recording of brass and voice. Table II shows output levels requiring 10-20 dB of padding under
these conditions, and this does agree with recording practice today where solid-state amplifiers are
used. But most tube consoles did not have input pads and yet the same microphone performed with
little noticeable distortion. Certainly brass players and singers are not that much louder today than
they were yesterday. The microphone distance is about the same. The preamplifier specifications
have not changed that much. Yet transistors require pads and tubes do not.
Table 2. Peak output for a U-47 microphone for various sounds.
Instrument
Peak
Pressure Microphone
Distance Frequency Voltage (db
(feet)
(Hz)
Ref. 0.775 V)
75-piece orchestra
15
350
-10
15-piece orchestra
10
350
-12
Trumpet
3
600
-16
Trombone
3
600
-15
French horn
3
300
-13
Flute
3.5
800
-26
Piccolo
3.5
2500
-18
Clarinet
3.5
350
-22
Bass sax
3.5
350
-8
Bass viol
5
150
-13
Here then is the hypothesis for further investigation. In the usual evaluation of audio preamplifiers it
is assumed that they are operated in their linear range, i.e., harmonic distortion less than 10%. In this
range tubes and transistors do have very similar performance characteristics. But the preceding
section points out that amplifiers are often operated far out of their linear range at signal levels which
would cause severe distortion. Under these conditons, tubes and transistors appear to behave quite
differently from a sound viewpoint.
DISTORTION CHARACTERISTICS OF PREAMPLIFIERS
Three commercially available microphone preamplifiers of different designs were set up in the
recording studio. Each amplifier was adjusted for a gain of 40 dB and an overload point of 3% total
harmonic distortion (THD) at +18 dBm. Preamplifier 1 was a transistor design, preamplifier 2 was a
hybrid operational amplifier, and preamplifier 3 was a vacuum-tube triode design. The amplifier
outputs were terminated in 600-ohm loads and bridged by the monitoring system. The test signal,
U-87 microphone, and large floor tom were switchable to each preamplifier input.
An informal group of studio personnel listened to the outputs of the three amplifiers on the normal
control room monitor speakers. As the test signal was switched from one amplifier to another,
listeners were asked to judge the sound quality. The output of amplifiers 1 and 2 was unanimously
judged to be severely distorted. Amplifier 3, however, sounded clean. The test was repeated several
times inserting attenuating pads in the microphone line until each amplifier sounded undistorted.
Amplifier 1 could stand overloads of 5-10 dB without noticeable distortion. Amplifier 2 showed
noticeable distortion at about 5 dB overload. Further listening revealed that it was only in the range
of early overload where the amplifiers differed appreciably in sound quality. Once the amplifiers
were well into the distortion region, they all sounded alike -- distorted. In their normal nonoverload
range all three amplifiers sounded very clean.
The listening tests clearly indicate that the overload margin varies widely between different types of
amplifiers. Engineering studios show that any amplifier adds distortion as soon as the overload point
is reached. The tests show that all amplifiers could be overloaded to a certain degree without this
distortion becoming noticeable. It may be concluded that these inaudible harmonics in the early
overload condition might very well be causing the difference in sound coloration between tubes and
transistors.
To get a general representation of the character of harmonic distortion in audio amplifiers, overload
curves were plotted for about fifty different circuits. The tube circuits used the popular 12AY7 and
12AX7 triodes, the 8628 and 7586 triode nuvistors, and the 5879 pentode. These tubes have all been
extensively used in recording console preamplifiers. The 2N3391A, 2N5089, and 2N3117 silicon
PNP transistors were also chosen because of their extensive use in console and tape recorder
circuitry. For comparison purposes tests were also run on the 2N5087 which is the PNP sister of the
2N5089. Operational amplifiers included the popular 709 and LM301 monolithic units and two
commercially available hybrid designs used in recording consoles.
Fig. 2. Single-stage amplifier comparison of total harmonic distortion (THD).
The curves shown in Fig. 2 are representative of the general distortion characteristics of single-stage
class A audio amplifiers. The devices are all operating open loop (no feedback) with a bias point
which allows for maximum undis- torted output swing. The curves are referenced to a common
point of 3% (THD), regardless of actual input or output levels. Since the objective of these
comparisons is to detect variations in the slopes of the distortion character- istics, the x axis is a scale
of relative level independent of circuit impedance considerations. These particular curves were
chosen from the many plotted as representative of different families: silicon transistors, triodes, and
pentode. A quick look shows that the often versed opinion that tubes overload more gently than
transistors is obviously a myth.
Fig. 3. Multistage amplifier comparison of total harmonic distortion (THD).
Fig. 3 shows the distortion characteristics for four differ- ent commercially available preamplifiers,
using two or more stages of amplification. All the circuits use feedback, a couple are push-pull. Each
amplifier is operating into 600 ohms at a gain of 40 dB. As in the previous curves, there is a
common reference point of 1% THD. While these curves show a marked difference from the
single-stage amplifiers, a review of the many different amplifiers tested shows that the slopes of all
THD curves run about the same. The lack of a wide variation between the curves indicates that THD
plots are not very relevant to what the ear hears in the listening tests.
Another series of tests were made on the same group of preamplifiers using a spectrum analyzer to
measure the amplitude of individual harmonics. Each amplifier was driven 12 dB into overload,
starting from a reference point of 1% third harmonic distortion. Every harmonic to the seventh was
plotted. Since it is not possible to measure the relative phase of the harmonics on the spectrum
analyzer, the over- load waveforms were recorded for Fourier analysis on the digital computer. The
resulting plots divided amplifiers into three distinct categories.
1) Tube Characteristics
Fig. 4. Distortion components for two-stage triode amplifier.
Fig. 4 shows the distortion components for a typical two- stage 12AY7 amplifier. This particular
design is quite representative of several single-ended, multistage triode tube amplifiers tested. The
outstanding characteristic is the dominance of the second harmonic followed closely by the third.
The fourth harmonic rises 3-4 dB later, running parallel to the third. The fifth, sixth, and seventh
remain below 5% out to the 12-dB overload point. These curves seem to be a general characteristic
of all the triode amplifiers tested, whether octal, miniature, nuvistor, single-ended, or push-pull. Fig.
5 is the waveform at 12 dB of overload. The clipping is unsymmetrical with a shifted duty cycle.
Again this is a characteristic of all the triode amplifiers tested.
Fig. 5. Waveform of triode amplifier of Fig. 4 at 12-dB overload. 1000-Hz tone
Fig. 6. Distortion components for two-stage pentode amplifier.
Fig. 6 shows the distortion components for a two-stage single-ended pentode amplifier. Here the
third harmonic is dominant and the second rises about 3 dB later with the same slope. Both the
fourth and fifth are prominent while the sixth and seventh remain under 5%. The waveform at 12-dB
overload (Fig. 7), is similar to the triode, but its duty cycle is not shifted as much. It is not reasonable
to assume that virtually all tube amplifiers can be represented by these two examples. However, the
major characteristic of the tube amplifier is the presence of strong second and third harmonics,
sometimes in concert with the fourth and fifth, but always much greater in amplitude. Harmonics
higher than the fifth are not significant until the overload is beyond 12 dB. These characteristics
seem to hold true for wide variations in circuit design parameters. The extreme difference in the tube
amplifiers is the interchanging of the position of the second and third harmonics. This effect is not
just a characteristic of the pentode, it is common to triodes too.
Fig. 7. Waveform of pentode amplifier of Fig. 6 at 12 dB overload, 1000-Hz tone.
Fig. 8. Distortion components for multistage capacitor-coupled transistor amplifier.
2) Transistor Characteristics
Figs. 8 and 10 show the characteristics of two transistor amplifiers. Like the previous figures the
curves are representative of all the transistor amplifiers tested. The distinguishing feature is the
strong third harmonic component. All other harmonics are present, but at a much lower amplitude
than the third. When the overload reaches a break point, all the higher harmonics begin to rise
simultaneously. This point is generally with 3-6 dB of the 1% third harmonic point. The waveforms
of these amplifiers (Figs. 9 and 11) are distinctly square wave in form with symmetrical clipping and
an almost perfect duty cycle. Both amplifiers shown have single-ended inputs and push-pull outputs.
However, the circuit designs are radically different.
Fig. 9. Waveform for transistor amplifier of Fig. 8 at 12-dB overload, 1000-Hz tone.
Fig. 10. Distortion components for multistage transformer-coupled transistor amplifier.
Fig. 11. Waveform for transistor amplifier of Fig. 10 at 12-dB overload, 1000-Hz tone.
3) Operational-Amplifier Characteristics
Fig. 12 is a hybrid operational amplifier. The third harmonic rises steeply as the dominant distortion
component in a characteristic similar to the transistor. Also rising very strongly from the same point
are the fifth and seventh harmonics. All even harmonics are suppressed completely. The waveform
of Fig. 13 is a perfect square wave. As a classification group, operational amplifiers have the most
uniform characteristics with almost no deviation from the curves shown in this example.
Fig. 12. Distortion components for monolithic operational amplifier with hybrid output stage.
Fig. 13. Waveform for operational amplifier of Fig. 12 at 12-dB overload, 1000-Hz tone.
In view of the transient nature of audio signals, steady- state single-frequency distortion analysis
could yield questionable results. Indeed, the arguments for and against sine-wave and pulse test
signals for audio system testing have been the subject for a number of engineering papers [4], [7].
For our purposes, however, a few minutes toying with an electronic synthesizer quickly proved that
musical instruments do not produce fast pulses. For example, a good simulation of the large floor
tom used in the amplifier listening tests is a 100-Hz tone modulated with an envelope rise time of 5
ms and a decay time of 300 ms. Also an extensive study of trumpet tones [6] measured the rise time
of the fastest staccato notes at 12 ms. Certainly, rise times of these orders can not be considered
pulses for audio amplifiers with passbands extending to 20 kHz or better. Just to further prove the
correctness of the preceding steady-state results, the synthesized floor tom signal was used to test the
same amplifiers at the same level as the microphone signal.
Fig. 14. a. Envelope of Moog-generated floor tom test signal. b. Envelope clipping of transient
signals by amplifier is identical to single-frequency clipping levels.
Careful observation of the amplified signal showed that envelope clipping was identical to the
steady-state clipping level (Fig. 14). There were no glitches or other fast transient phenomena in the
output signal.
SIGNIFICANCE OF MUSICAL HARMONICS
Having divided amplifiers into three groups of distortion characteristics, the next step is to determine
how the harmonics relate to hearing. There is a close parallel here between electronic distortion and
musical tone coloration that is the real key to why tubes and transistors sound different. Perhaps the
most knowledgeable authorities in this area are the craftsman who build organs and musical
instruments [8], [9]. Through many years of careful experimentation these artisans have determined
how various harmonics relate to the coloration of an instrument's tonal quality.
The primary color characteristic of an instrument is determined by the strength of the first few
harmonics. Each of the lower harmonics produces its own characteristic effect when it is dominant
or it can modify the effect of another dominant harmonic if it is prominent. In the simplest
classification, the lower harmonics are divided into two tonal groups. The odd harmonics (third and
fifth) produce a "stopped" or "covered" sound. The even harmonics (second, fourth, and sixth)
produce "choral" or "singing" sounds.
The second and third harmonics are the most important from the viewpoint of the electronic
distortion graphs in the previous section. Musically the second is an octave above the fundamental
and is almost inaudible; yet it adds body to the sound, making it fuller. The third is termed quint or
musical twelfth. It produces a sound many musicians refer to as "blanketed." Instead of making the
tone fuller, a strong third actually gives the sound a metallic quality that gets annoying in character as
its amplitude increases. A strong second with a strong third tends to open the "covered" effect.
Adding the fourth and fifth to this changes the sound to an "open horn" like character.
The higher harmonics, above the seventh, give the tone "edge" or "bite." Provided the edge is
balanced to the basic musical tone, it tends to reinforce the fundamental, giving the sound a sharp
attack quality. Many of the edge harmonics are musically unrelated pitches such as the seventh,
ninth, and eleventh. Therefore, too much edge can produce a raspy dissonant quality. Since the ear
seems very sensitive to the edge harmonics, controlling their amplitude is of paramount importance.
The previously mentioned study of the trumpet tone [6] shows that the edge effect is directly related
to the loudness of the tone. Playing the same trumpet note loud or soft makes little difference in the
amplitude of the fundamental and the lower harmonics. However, harmonics above the sixth
increase and decrease in amplitude in almost direct proportion to the loudness. This edge balance is a
critically important loudness signal for the human ear.
RELATIONSHIP OF FACTORS AND FINDINGS
The basic cause of the difference in tube and transistor sound is the weighting of harmonic distortion
components in the amplifier's overload region.
Transistor amplifiers exhibit a strong component of third harmonic distortion when driven into
overload. This harmonic produces a "covered" sound, giving a recording a restricted quality.
Alternatively a tube amplifier when overloaded generates a whole spectrum of harmonics.
Particularly strong are the second, third, fourth, and fifth overtones, which give a full-bodies
"brassy" quality to the sound. The further any amplifier is driven into saturation, the greater the
amplitude of the higher harmonics like the seventh, eighth, ninth, etc. These add edge to the sound
which the ear translates to loudness information. Overloading an operational amplifier produces such
steeply rising edge harmonics that they become objectionable within a 5-dB range. Transistors
extend this overload range to about 10 dB and tubes widen it to 20 dB or more. Using this basic
analysis, the psychoacoustic characteristics stated in the beginning of this paper can be related to the
electrical harmonic properties of each type of amplifier.
It was not part of the original intent of this paper to analyze operational amplifiers. However, the
tests show that they fall into a distinct class of their own. Basically, operational amplifiers produce
strong third, fifth and seventh harmonics when driven only a few dB into overload. The resultant
sound is metallic with a very harsh edge which the ear hears as strong distortion. Since this sound is
so objectionable, it acts as a clearly audible overload warning signal. Consequently, operational
amplifiers are rarely operated in their saturation region. This results in a very cleanly amplified sound
with little coloration and true dynamic range within the limitations of the amplifier. True dynamic
range is not necessarily the determinant of good sound reproduction, however, since it is much
greater than any disc or tape system presently available. Because of their characteristics, operational
amplifiers produce only the top end of the dynamic range which contains all the transients but lacks
the solid pitch information which the ear hears as music. When records of true dynamic range are
played on a limited-range system, they sound very thin. This relates directly to the originally cited
listener's comment that transistor records were very clean but sounded sibilant and cymbally.
The transistor characteristics which our subjects noted were the buzzing or white-noise sound and
the lack of "punch." The buzz is of course directly related to the edge produced by overloading on
transients. The guess that this is white noise is due to the fact that many of the edge harmonics like
the seventh and ninth are not musically related to the fundamental. The ear hears these dissonant
tones as a kind of noise accompanying every attack. The lack of punch is due to the strong third
harmonic which is inaudibly "blanketing" the sound. This is correctable by using a large enough pad
to prevent all peaks from reaching the amplifier's saturated region. But from a practical standpoint,
there is no way of determining this on most consoles. Adding auxiliary peak indicators on the input
preamplifiers could alleviate both these problems, and the sound would be very close to that of the
operational amplifier in its linear region.
Vacuum-tube amplifiers differ from transistor and operational amplifiers because they can be
operated in the overload region without adding objectionable distortion. The combination of the slow
rising edge and the open harmonic structure of the overload characteristics form an almost ideal
sound- recording compressor. Within the 15-20 dB "safe" overload range, the electrical output of the
tube amplifier increases by only 2-4 dB, acting like a limiter. However, since the edge is increasing
within this range, the subjective loudness remains uncompressed to the ear. This effect causes
tube-amplified signals to have a high apparent level which is not indicated on a volume indicator
(VU meter). Tubes sound louder and have a better signal-to-noise ratio because of this extra
subjective head room that transistor amplifiers do not have. Tubes get punch from their naturally
brassy overload characteristics. Since the loud signals can be recorded at higher levels, the softer
signals are also louder, so they are not lost in tape hiss and they effectively give the tube sound
greater clarity. The feeling of more bass response is directly related to the strong second and third
harmonic components which reinforce the "natural" bass with "synthetic" bass [5]. In the context of
a limited dynamic range system like the phonograph, recordings made with vacuum-tube
preamplifiers will have more apparent level and a greater signal to system noise ratio than recordings
made with transistors or operational amplifiers.
ACKNOWLEDGMENT
The author wishes to thank Walter Sear and Peter Scheiber for innumerable helpful discussions on
the musician's viewpoint of sound. He also wishes to thank John Olson of RCA and Steve Temmer
of Gotham Audio for the loaning of amplifiers
REFERENCES
[1] "Neumann Transistor Condensor Microphones," Gotham Audio Corp., Sales Bulletin (1971).
[2] A. D. Smith and P. H. Wittman, "Design Considerations of Low-Noise Audio Input Circuitry
for a Professional Microphone Mixer," J. of the Audio Eng. Soc., vol. 18, pp. 140-156 (April 1970).
[3] A. Schaumberger, "The Application of Impulse Measurement Techniques to the Detection of
Linear Distortion," J. of the Audio Eng. Soc., vol. 19, pp. 664-668 (Sept. 1971).
[4] M. Otala, "Circuit Design Modifications for Minimizing Transient Intermodulation Distortion in
Audio Amplifiers," J. of the Audio Eng. Soc., vol. 20, pp. 396-399 (June 1972)
[5] F. Langford-Smith, Radiotron Designer's Handbook (RCA, 1953), chapter 14.
[6] J. C. Risset, "Computer Study of Trumpet Tones," Bell Telephone Laboratories, File
MM-66-1222-2.
[7] J. R. Ashley, T. A. Saponas, and R. C. Matson, "Test Signals for Music Reproduction
Systems," IEEE Spectrum, vol. 8, pp. 53-61 (July 1971).
[8] A. H. Benade, Horns, Strings and Harmony, (Doubleday, New York, 1960).
[9] R. A. Schaefer, "New Techniques for Organ Tone Generation," J. of the Audio Eng. Soc., vol.
19, pp. 570-575 (July/Aug. 1971).
[10] R. Langevin, "Intermodulation Distortion in Tape Recording,"J. of the Audio Eng. Soc., vol. 11,
pp. 270-278 (July 1963).
THE AUTHOR
Russell O. Hamm received his engineering training at the University of New Hampshire. He worked
for Vidcom Electronics and later the Fine Recording division of that company, designing and
supervising the installation of their extensive 16-track recording facilities. While with Fine
Recordings, Mr. Hamm did a great deal of experimentation in stereophonic and quadraphonic sound
for records and motion pictures which, in conjunction with Peter Scheiber, formed the basis for the
development of the present matrix-quad record. Mr. Hamm's record-producing and engineering
credits include albums, commercials, and motion-picture sound tracks by many well-known artists.
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