From: gabe@pgm.com (Gabe M. Wiener)

Newsgroups: rec.audio.pro,rec.answers,news.answers
Subject: rec.audio.pro FAQ (v 0.9)

Frequently Asked Questions (FAQ) file for rec.audio.pro
Version 0.8


Section II - The business of audio

Q2.1 - How does one get started as a professional audio engineer?
Q2.2 - Are audio schools worth the money? Which schools are best?
Q2.3 - What are typical rates for various professional audio services?

Section III - Audio Interconnections

Q3.1 - How are professional transmission lines and levels different from
consumer lines and levels? What is -10 and +4? What's a balanced
or differential line?
Q3.2 - What is meant by "impedance matching"? How is it done? Why is it necessary?
Q3.3 - What is the difference between dBv, dBu, dBV, dBm, dB SPL, and plain old dB?
Q3.4 - Which is it for XLRs? Pin 2 hot? Or pin 3 hot?
Q3.5 - What is phantom power? What is T-power?
Q3.6 - How do I interconnect balanced and unbalanced components?
Q3.7 - What are ground loops and how do I avoid them?
Q3.8 - What is the "Pin 1 problem" and how do I avoid it?

Section IV - Analog tape recording

Q4.1 - What does it mean to "align" a tape machine?
Q4.2 - What is bias? What is overbias?
Q4.3 - What is the difference between Dolby A, B, C, S, and SR? How do each of these systems work?
Q4.4 - What is Dolby HX-Pro?
Q4.5 - How does DBX compare to Dolby?
Q4.6 - How much better are external microphone preamplifiers than those found in my portable recorder?
Q4.7 - What is an MRL? Where do I get one?

Section V - Digital recording and interconnection

Q5.1 - What is sampling? What is a sampling rate?
Q5.2 - What is oversampling?
Q5.3 - What is the difference between a "1 bit" and a "multibit" converter? What is MASH, what is Delta/Sigma? Should I really care?
Q5.4 - On an analog recorder, I was always taught to make sure the signal
averages around 0 VU. But on my new DAT machine, 0 is all the way at
the top of the scale. What's going on here?
Q5.5 - Why doesn't MiniDisc or Digital Compact Cassette sound as good as DAT or CD? After all, they're both digital.
Q5.6 - What is S/P-DIF? What is AES/EBU?
Q5.7 - What is clock jitter?
Q5.8 - How long can I run AES/EBU or S/P-DIF cables? What kind of cable should I use?

Q5.9 - What is SCMS? How do I defeat it?!

Q5.10 - What is PCM-F1 format?
Q5.11 - How do digital recorders handle selective synchronization?
Q5.12 - How can a 44.1 kHz sampling rate be enough?
Q5.13 - Doesn't the 44.1 kHz sampling rate make it impossible to reproduce square waves?
Q5.14 - How can a 16-bit word-length be enough?
Q5.15 - What's all this about 20- and 24-bit digital audio? Aren't
CDs limited to 16 bits?

Section VI - Digital editing and mastering

Q6.1 - What is a digital audio workstation?
Q6.2 - How is digital editing different from analog editing?
Q6.3 - What is mastering?
Q6.4 - What is normalizing?
Q4.5 - I have a fully edited DAT that sounds just like I want it to sound on the CD.
Q6.6 - What is PCM-1630? What is PMCD?
Q6.7 - When preparing a tape for CD, how hot should the levels be?
Q6.8 - Where can I get CDs manufactured?
Q6.9 - How are CD error rates measured, and what do they mean?

Section VII - Market survey. What are my options if I want --

Q7.1 - A portable DAT machine
Q7.2 - A rack size DAT machine
Q7.3 - An inexpensive stereo microphone
Q7.4 - An inexpensive pair of microphones for stereo
Q7.5 - A good microphone for recording vocals
Q7.6 - A good microphone for recording [insert instrument here]
Q7.7 - A a small mixer
Q7.8 - A portable cassette machine
Q7.9 - A computer sound card for my IBM PC or Mac
Q7.10 - An eight-track digital recorder?

Section VIII - Sound reinforcement

Q8.1 - We have a fine church choir, but the congregation can't hear them. How do we mic the choir?
Q8.2 - How do I 'ring out' a system?
Q8.3 - How much power to I need for [insert venue here]?
Q8.4 - How good is the Sabine feedback eliminator?

Section IX - Sound restoration

Q9.1 - How can I play old 78s?
Q9.2 - How can I play Edison cylinders?
Q9.3 - What are "Hill and Dale" recordings, and how do I play them back?
Q9.4 - What exactly are NoNOISE and CEDAR? How are they used?
Q9.5 - How do noise suppression systems like NoNOISE and CEDAR work?
Q9.6 - What is forensic audio?

Section X - Recording technique, Speakers, Acoustics, Sound

Q10.1 - What are the various stereo microphone techniques?
Q10.2 - How do I know which technique to use in a given circumstance?
Q10.3 - How do I soundproof a room?
Q10.4 - What is a near-field monitor?
Q10.5 - What are the differences between "studio monitors" and home loudspeakers?

Section XI - Industry information

Q11.1 - Is there a directory of industry resources?
Q11.2 - What are the industry periodicals?
Q11.3 - What are the industry trade organizations?
Q11.4 - Are there any conventions or trade shows that deal specifically with professional audio?

Section XII - Miscellaneous

Q12.1 - How do I modify Radio Shack PZMs?
Q12.2 - Can I produce good demos at home?
Q12.3 - How do I remove vocals from a song?

Section XIII - Bibliography

Q13.1 - Fundamentals of Audio Technology
Q13.2 - Studio recording techniques
Q13.3 - Live recording techniques
Q13.4 - Digital audio theory and practice
Q13.5 - Acoustics
Q13.6 - Practical recording guides

Section XIV

Q14.1 - Who wrote the FAQ
Q14.2 - How do you spell and pronounce the FAQ maintainer's surname?


Section II - The business of audio

Q2.1 - How does one get started as a professional audio engineer?

There are as many getting-started stories as there are audio
engineers. The routes into the industry are highly dependent on what
aspect of the industry one wishes to enter. For instance, many
engineers who work in the classical-music field have at one time or
another been classical performers. Others enter through their work in
other musical genres, or through engineering programs at universities
or technical schools. Without exception, everyone in the industry has
learned at least a portion of their craft from watching those with
more hands-on experience. Whether this comes from a formal internship
or just from sustained observation and long-term question-asking, it
is almost always universally true. [Gabe]

Q2.2 - Are audio schools worth the money? Which schools are best?

An audio school will teach you the basics of the audio business,
but just like any technical school, what they teach you may not be
worth what you pay.

There are several schools of thought:

1. Audio schools are great, you get trained on the gear that is used
by top studios and costs millions of dollars, you get taught by pros
in the field and you have job placement assistance after you graduate.

2. Going to an audio school is like wanting to learn aviation,
and when you start flight school they teach you a 747. In the
real world, you are probably not going to have 96 channel automated
consoles on your first job. You are not going to mix your first
live gig on a 48-channel 100,000 watt stadium PA rig. Better to
start off on real world equipment and work your way up to the
top-of-the-line stuff. Most recording studios are 24-track analog
or less and most PA systems are 16 channel, 3,000 watts or less.
Don't buy education for something you will never get to use after
you leave the school.

3. Audio Schools are a waste of money. Instead of spending $18,000
for a course and having nothing to show for it but a technical
certificate (which everyone knows is no help at all getting a job),
you would be better off spending the 18 grand on books and gear and
learning by trial and error, or saving the 18 grand altogether and
learning first from reading, and later from apprenticing.
[jsaurman@cftnet.com (Jim Saurman)]

Jim summarizes the opinions pretty well. Recognize that an altogether
different option is to attend a full four-year college program. Many
colleges and universities offer such programs. Examples include
Peabody Conservatory, Cleveland Institute of Music, McGill University,
New York University, University of Miami at Coral Gables, and the
University of Massachusetts at Lowell. Without fail, graduates from
these sorts of programs earn far more respect than graduates of any
technical school. [Gabe]

Q2.3 - What are typical rates for various professional audio services?

Depends on what you want to have done, and where.

One can pay upwards of $300/hr for prime studio rental time in New York.
In a small community however, one might find a project studio for $25/hr.
Generally speaking, the rule is: the rarer the service, the more it will
cost. In a community with dozens of small 8-track studios, you won't
have to pay much. If you need emergency audio restoration, or mastering
by a top-flight pop-music engineer, you can expect to drop many hundreds
of dollars an hour. Like so many other things in this industry, there
are no rules, and Smith's invisible hand guides the market. [Gabe]

Section III - Audio Interconnections

Q3.1 - How are professional transmission lines and levels different from
consumer lines and levels? What is -10 and +4? What's a balanced
or differential line?

Professional transmission lines differ from consumer lines in two
ways. First, consumer lines tend to run about 14 dB lower in level
than pro lines. Second, professional lines run in differential, or
balanced, configuration.

In a single-ended line, the signal travels down one conductor and
returns along a shield. This is the simplest form of audio
transmission, since it is essentially the same AC circuit you learned
about in high-school physics. The problem here is that any noise
or interference that creeps into the line will simply get added to
the signal and you'll be stuck with it.

In a differential line, there are three conductors. A shield, a
normal "hot" lead, and a third lead called the "cold" or "inverting"
lead, which carries a 180-degree inverted copy of the hot lead. Any
interference that creeps into the cable thus affects both the hot and
cold leads equally. At the receiving end, the hot and cold leads are
summed using a differential amplifier, and any interference that has
entered the circuit (called "common-mode information" since it is
common to both the hot and cold leads), gets canceled out.
Differential lines are thus better suited for long runs, or for
situations where noise or interference may be a factor. [Gabe]

Q3.2 - What is meant by "impedance matching"? How is it done? Why is it

We can talk about the characteristic impedance of an input, which is to
say the ratio of voltage to current that it likes to see, or how much
it loads down a source. (You can think of this as being an "AC resistance"
and you would be mostly right, although it's actually the absolute
magnitude of the vector drawn by the resistive and reactive load
components. Dealing with line level signals, reactive components
are going to be negligible, though).

In general, in this modern world, most equipment has a low impedance
output, going into relatively high impedance input. This wastes some
amount of power, but because electricity is cheap and it's possible to
build low-Z outputs easily today, this is not a big deal.

With microphones, it _is_ a big deal, because the signal levels are
very low, and the drive ability poor. As a result, we try and get the
best efficiency possible from microphones to get the lowest noise
floor. This is often done by using transformers to step up the voltage
or step it down, to go into a higher or lower Z load. Transformers
have some major disadvantages in that they can be significant sources
of nonlinearity, but back in the days of tubes they were the only
solution. Tubes have a very high-Z input, and building balanced inputs
with tubes requires three devices instead of one. As a result, all
mike preamps would have a 600 ohm balanced input, with a transformer,
driving a preamp tube. Today, transistor circuits can be used for
impedance matching, although they are often more costly and can be noisier
in cases.

As a result of the expense, consumer equipment was built with high-Z
microphone inputs, and high-Z microphones. This resulted in more noise
pickup problems, but was cheaper to make. Unfortunately this still
held on into the modern day of the transistor, and a lot of high-Z
consumer gear exists. Guitar pickups are generally high-Z devices,
and require a direct box to reduce the impedance so that they can go into
a standard 600 ohm mike preamp directly.

Many years ago, the techniques that were used in audio came originally
from telephone company practice. Phone systems operate with 150 or 600
ohm balanced lines, and adoption of this practice into the audio industry
caused those standards to be used. In the modern age where lines are
relatively short and transformers considered problematic, the tendency
has been to have low-Z outputs for all line level devices, driving
high-Z inputs. While this is not the most efficient system, it is relatively
foolproof, and appears on most consumer equipment. A substantial amount of
professional gear, however, still uses internal balancing transformers or
resistor networks to match to a perfect 600 ohm impedance. [Scott]

[Ed. note: Modern equipment works on principles of voltage transfer
rather than power transfer. Thus a standard audio circuit today is
essentially a glorified voltage divider. You have a very low output
impedance and a very high input impedance such that the most voltage
is dropped across the load. This is not an impedance-matched circuit
in the classic sense of the word. Rather, it is a "bridged" or
"constant voltage" impedance match, and is the paradigm on which
nearly all audio circuits operate nowadays. -Gabe]

Q3.3 - What is the difference between dBv, dBu, dBV, dBm, dB SPL, and plain
old dB? Why not just use regular voltage and power measurements?

Our ears respond logarithmically to increases in sound pressure level.
In order to simplify the calculations of these levels, as well as the
electrical equivalents of them in audio systems, the industry uses a
logarithmic system to denote the values. Specifically, the decibel is
used to denote logarithmic level above a given reference. For
instance, when measuring sound pressure level, the basic reference
against which we take measurements is the threshold of hearing for
the average individual, 10^-12 W/m^2. The formula for dB SPL then

10 Log X / 10^-12 where X is the intensity in W/m^2

The first people who were concerned about transmitting audio over
wires were, of course, the telephone company. Thanks to Ma Bell we
have a bunch of other decibel measurements. We can use the decibel to
measure electrical power as well. In this case, the formula is
referenced to 1 milliwatt in the denominator, and the unit is dBm. 1
milliwatt was chosen as the canonical reference by Ma Bell. Since
P=V^2 / R, we can also express not only power gain in dB but also
voltage gain. In this case the equation changes a bit, since we have
the ^2 exponent. When we take the logarithm, the exponent comes
around into the coefficient, making our voltage formula 20 log.
In the voltage scenario, the reference value becomes 0.775 V (the
voltage drop across 600 ohms that results in 1 mW of power). The
voltage measurement unit is dBv.

The Europeans, not having any need to abide by Ma Bell's choice for a
canonical value, chose 1V as their reference, and this is reflected
as dBV instead of dBv. To avoid confusion, the Europeans write the
American dBv as dBu. Confused yet? [Gabe]

Q3.4 - Which is it for XLRs? Pin 2 hot? Or pin 3 hot?

Depends on whom you ask! Over the years, different manufacturers have
adopted varying standards of pin 2 hot and pin 3 hot (and once in a
while, pin *1* hot!). But nowadays most manufacturers have adopted
pin 2 hot. Still, it is worth taking the extra minute or two to check
the manual. The current AES standard is pin 2 hot. [Gabe]

Q3.5 - What is phantom power? What is T-power?

Condenser microphones have internal electronics that need power
to operate. Early condenser microphones were powered by
batteries, or separate power supplies using multi-conductor
cables. In the late 1960's, German microphone manufacturers
developed 2 methods of sending power on the same wires that carry
the signal from the microphone.

The more common of these methods is called "phantom power" and is
covered by DIN spec 45596. The positive terminal of a power
supply is connected through resistors to both signal leads of a
balanced microphone, and the negative terminal is connected to
ground. 48 volts is the preferred value, with 6800 ohm resistors
in each leg of the circuit, but lower voltages and lower resistor
values are also used. The precise value of the resistors is not
too critical, but the two resistors must be matched within 0.4%.

Phantom power has the advantage that a dynamic or ribbon mic may
be plugged in to a phantom powered microphone input and operate
without damage, and a phantom powered mic can be plugged in to
the same input and receive power. The only hazard is that in case
of a shorted microphone cable, or certain old microphones having
a grounded center tap output, current can flow through the
microphone, damaging it. It's a good idea anyway to check cables
regularly to see that there are no shorts between any of the
pins, and the few ribbon or dynamic microphones with any circuit
connection to ground can be identified and not used with phantom

T-power (short for Tonaderspeisung, also called AB or parallel
power, and covered by DIN spec 45595) was developed for portable
applications, and is still common in film sound equipment.
T-power is usually 12 volts, and the power is connected across
the balanced pair through 180 ohm resistors. Only T-power mics
may be connected to T-power inputs; dynamic or ribbon mics may be
damaged and phantom powered mics will not operate properly. [David]

Q3.6 - How do I interconnect balanced and unbalanced components?

First, let's define what the terms mean. The simplest audio
circuit uses a single wire to carry the signal; the return path,
which is needed for current to flow in the wire, is provided
through a ground connection, usually through a shield around the
wire. This system, called unbalanced transmission, is very
susceptible to hum pickup and cannot be used for low level
signals, like audio, for more than a few feet. Balanced
transmission occurs when two separate and symmetrical wires are
used to carry the signal. A balanced input is sensitive only to
voltage that appears between the two input terminals; signals
from one terminal to ground are canceled by the circuit.

The simplest way to connect between balanced and unbalanced
equipment is to use a transformer. The signals are magnetically
coupled through the core of the transformer and either side may
be balanced or unbalanced. Good transformers are expensive,
however, and there are cheaper methods that can be used in some

An unbalanced output can be connected to a balanced input. For
instance, from the unbalanced output of a CD player, connect the
center pin to pin 2 of the balanced XLR input connector, and the
ground to pins 1 and 3. To connect the balanced output of something
to an unbalanced input requires different techniques depending on
whether the output is active balanced (each side has a signal with
respect to ground) or floating balanced (for instance, the secondary
of a transformer with no center-tap connection). If it's an active
balanced output, you can simply use half of it; connect pin 2 to the
unbalanced input, and pin 1 to ground, leaving pin 3 floating. If this
doesn't work (no or very weak signal) connect pin 3 of the output to
pin 1 and ground and leave pin 2 connected to the unbalanced input
center pin. Some active balanced outputs, particularly microphones,
use the balanced circuit to cancel distortion, so this hookup may
result in higher distortion than if a proper balanced-to-unbalanced
converter such as a differential stage or a transformer were used.

Q3.7 - What are ground loops and how do I avoid them?

One of the most difficult troubleshooting tasks for the audio
practitioner is finding the source of hum, buzz and other
interfering signals in the audio signal. Often these are caused
by "ground loops." This unfortunate and inaccurate term (it need
not be in the "ground" path, and the "loop" is not what causes
the problem) is poorly understood by most users of audio
equipment. A better name for this phenomenon is "shared path
coupling" because it happens when two signals share the same path.

Another semantic problem that should be addressed early on is the
idea that "ground" is one place where all currents go. It's not,
there's nothing special about calling a signal "ground," current
still flows through any path that's available to it.

Referring to the discussion above regarding unbalanced signal
paths, recall that there must be a complete circuit from the
output of some device, through the input of another device and
back to the "return" side of the output if any current is to
flow. Current doesn't flow by itself, it must have a complete
path. If there are multiple paths over which the current might
flow, the current will be divided among them with most of the
current flowing through the path having the least resistance. Any
available path, regardless of the resistance in it, will carry
some of the current, it's not a case of all the current following
the path that has least resistance.

For example, suppose we have two units connected together through
a small piece of coaxial cable, and the units are also connected
together at the wall outlet through their grounded power cords --
the ground pins are connected to the chassis at each end. The
audio signal goes along the center of the coaxial cable, and part
of it might come back along the shield of the coax, but part will
also go through the ground wire of one unit and back through the
ground wire of the other unit. A problem arises when some other
signal is also flowing through this same return path. The other
signal might be another audio signal, video, data, or power. All
of the currents in a wire add together, and the resistance of the
wire causes a voltage to appear in proportion to the current
flowing. All of these voltages add together, so there is a little
bit of the video signal added to the audio, some of the power
signal added to the video, some of the power signal added to the
audio, etc. In rare instances, the "loop" of wire formed by the
intended ground return path and the happenstance lower resistance
return path formed by mounting hardware, power cords, etc. can
form a magnetic pickup as well, so that magnetic fields radiated
by transformers, CRT's, etc. can also induce a current in the
"loop," which makes yet another source of noise voltage.

This shared path coupling is a constant problem with unbalanced
audio systems. Lots of different methods have been tried to get
around the problem, many of them dangerous. Clipping off the
ground leads of equipment so there is no common power line path
between them simply makes any fault or leakage current follow
some other path, back through the signal cable to some equipment
that has a ground -- perhaps through the user's body, if all the
ground pins have been removed. The only general solution to
"ground loop" coupling with unbalanced equipment is to connect
all the chassis together with a very low resistance path (copper
strap or braid, for example), on the principle that since the
resistance is so low, any leakage current will produce a
correspondingly low signal voltage. It may also be effective to
interrupt the ground path of shield conductors over signal wires;
force the return path to go through the designated common strap
while leaving the shield in place only for electrostatic

With balanced equipment, no current should be flowing in the
shield conductors, and in fact performance should be identical
with the shield left disconnected at one end (preferably the
receiver end). Therefore balanced systems should be impervious to
shared path coupling or "ground loop" problems but in fact they
aren't, because most signals inside a given piece of equipment
are unbalanced, and there are often return paths internal to the
equipment that can be shared with return paths between other
units of equipment connected to it. Especially with mixed
digital, video and audio signals and high gain, high negative
feedback amplifier circuitry, this can be a big problem -- small
currents can create big effects -- and this brings us to the next
question. [David]

Q3.8 - What is the "Pin 1 problem" and how do I avoid it?

This is a special case of "ground loop" or shared path coupling.
Recently this has been discussed in great detail and clarity by a
group led by the consultant Neil Muncy of Toronto. Suppose you
have a mixer, whose balanced output is connected to an
amplifier's balanced input through a correctly wired cable. Both
units are powered from the AC mains and one or both have some
small amount of AC leakage current that travels to ground through
all available ground paths -- including the shield of the cable
that connects the two units. So far so good, no harm done because
the circuit is balanced and any common mode voltage from current
flowing through the shield will be canceled by the amplifier
input. However... a small part of this leakage current also
travels through the shield of the wire going from the back panel
XLR connector to the PC board, through some "ground" traces on
the PC board, and back out through the power line ground cable.
No problem so far, except that some gain stage on that same PC
board also uses that piece of ground trace in its negative
feedback loop, and some part of that leakage signal will be added
to the signal in that gain stage; it might be video, or data, or
another audio signal, or (most commonly) power.

The solution to this variant of shared path coupling is the same
sort of approach that applies to other unbalanced signals: give
the leakage current a very low resistance path to follow, and
remove as many of the shared paths as possible. Within a unit of
equipment, all the XLR connectors' pin 1 terminals should be
connected to ground with very low resistance (big) wire or
traces, and preferably all of the ground connections should be
made at one point, the so-called "star ground" system. A brute
force approach is to assume that the back panel is the star
ground, and wire every connector's pin 1 solidly to the panel as
directly as possible, and lift all the ground wires but one that
go from the connectors to the circuitry. In this way, all the
external leakage currents (the "fox" to use Neil Muncy's term)
will be conducted through the back panel and out of the way,
rather than running them through the ground traces on the PC
board where they will mix with internal low level signals in high
gain stages (the "hen house"). Individual wires can be run from
points on the circuit board that need to be at "ground" potential
to a common point on the back panel, which is designated a "zero
signal reference point" (ZSRP). Equipment that has a reputation
for being "quiet" and easy to use in many different applications
is often found to be wired this way, while equipment that is
"temperamental" if often found to be wired in such a way that
leakage currents are easily coupled to internal signal lines.

There's a simple test that can be done to check equipment
susceptibility to this problem. Connect the output, preferably
balanced and floating, of an ordinary audio oscillator to the pin
1 of any two XLR connectors on the equipment. Now operate the
equipment through its various modes, gain settings, etc. You may
be surprised to find the audio oscillator's signal appearing in
many different places in the equipment. [David]

Section IV - Analog tape recording

Q4.1 - What does it mean to "align" a tape machine?

There are a number of standard adjustments on any analogue tape
machine, which can roughly be broken up into mechanical and electronic
adjustments. The mechanical adjustments include the head position
(height, skew, and azimuth), and sometimes tape speed. Incorrect head
height will result in poor S/N and leakage between channels, because
the tracks on the head do not match up exactly with those on the tape.
Incorrect tape skew will result in level differences between channels
and uneven head wear, because there is more pressure on the top of the
head than the bottom (or vice versa). Incorrect azimuth will result
in loss of high frequency response and strange skewing of the stereo
image. Tape speed error will result in tonal shifts, although on many
machines with capstan speed controlled by crystal or line frequency,
it is not adjustable.

Electronic adjustments include level and bias adjustments for each
channel. Some machines may have bias frequency adjustments, equalization
adjustments for playback and record emphasis, pre-distortion adjustments,
and a varied bevy of adjustments for noise reduction systems.

Alignment is relatively simple, and the same general method applies
from the smallest cassette deck to the largest multitrack machine.
First, put a test tape on the machine. Use a real reference tape,
from the manufacturer, from MRL, or a similarly legitimate lab. DO
NOT EVER use a homebrew test tape that was recorded on a "known good"
machine. You will regret it someday. Spend the money and get a real
test tape (and not one of the flaky ones from RCA).

1. Speed adjustment (if necessary). Play back a 1 KHz reference tone
and, using a frequency counter, adjust the tape speed for proper
frequency output. There are strobe tapes available for this as well,
but with cheap frequency counters available, this method is much easier.

2. Head height and skew adjustments. Better see your machine's manual
on this one, because I have seen a variety of ways of doing this.

3. Azimuth adjustment. I find the easiest way to do this is to take
the left and right outputs and connect them to the X and Y inputs
of an oscilloscope, and play back a 1 KHz reference tone, while
adjusting the azimuth until a perfectly-diagonal line appears.
You can do this by ear if you are desperate, but I strongly recommend
the lissajous method, which is faster and more accurate. On multitrack
decks, use the two tracks as close as possible to the edge of the tape.
Now you have the playback head azimuth set... put a 1 KHz source into
the record input, with a blank tape on the machine, and adjust the
azimuth of the record head for the proper diagonal line.

4. Playback eq adjustment (if necessary). This is a case of playing
back various test tones at different frequencies, and adjusting the
response curve of the deck to produce a flat output. You can also
do this by playing back white noise and using a third-octave spectrum
analyzer of great accuracy to adjust for flat response. Again, this
is one to check your deck's manual for, because the actual
adjustments vary from one machine to another, and you will want to
use the test tape once again.

5. Record eq adjustment (if necessary). How this is done (and whether you
want to do it after biasing the tape) depends a lot on your deck.

6. Bias adjustment. There are a lot of ways to do this. My favorite method
is to use a white noise source, and adjust the bias until the source and
tape output sound identical. Some people prefer to use a signal generator
and set so that the levels of recorded tones at 1 KHz and 20 KHz are
identical. I find I can get within .5 dB by ear, though your mileage
may differ. [Ed. note: Many tapes have recommended overbias settings,
and many decks will also provide a chart that correlates the amount of
overbias against available tape formulations. -Gabe]

7. Record level adjustment. I use a distortion analyzer, and set the level
so that at +3 dB, I get 3% distortion on the output. Some folks who are
using very hot tape set the machines so that a certain magnetic flux is
produced at the heads given a certain input, but I find setting for
a given distortion point does well for me. If you don't have a distortion
analyzer, use a 1 KHz tone source and set so that you have the onset of
audible distortion at +3 dB, and you will be extremely close.

[Ed. note: The traditional way to do this is to align the repro side
of the machine using a calibration tape, and then to put the machine
into record. Monitoring off the repro head, the operator then aligns
the record electronics until the output is flat. -Gabe]

At this point, you will be pretty much set. Whether you want to do this
all on a regular basis is a good question. You should definitely go
through the complete procedure if you ever change brands of tape. Checking
the mechanical parameters on a regular basis is a good idea with some
decks (like the Ampex 350), which tend to drift. Clean your heads
every time you put a new reel on, and demagnetize regularly. [Scott]

Q4.2 - What is bias? What is overbias?

With just the audio signal applied to a tape, the frequency response
is very poor. High frequency response is much better than low
frequency, and the low frequency distortion is very high. In 1906,
the Poulson Telegraphone managed to record an intelligible voice on a
magnetic medium, but it was not until the 1930s when this problem was
solved by German engineers.

To compensate for the tape characteristic, a very high frequency
signal is applied to the tape in addition to the audio. This is
typically in the 100 KHz range, far above the audio range. With the
bias adjusted properly, the frequency response should be flat across
the audible range. With too low bias, bass distortion will be the
first audible sign, but with too much bias, the high frequency
response will drop off.

Incidentally, digital recording equipment takes advantage of the very
nonlinearity that is a problem with analogue methods. It records a
square wave on the tape, driving the tape into saturation at all
times, and extracts the signal from the waveform edges. As a result,
no bias is required. (For a good example of the various digital
recording methods, check out NASA SP 5038, _Magnetic Tape Recording_.)

[Ed. note: For those looking for an understanding of why we need
bias in the first place, here is one way to think about it. Tape
consists of lots of small magnetic particles called domains. These
domains are exposed to a magnetic field from the record head and
oscillate in polarity as the AC signal voltage changes. Domains,
being physical objects, have inertia. Every time the analog signal
crosses from positive to negative and back again, the voltage passes
the zero point for an instant. At this moment, the domain is at rest,
and like any other physical object, there is a short period of inertia
before it gets moving again. The result is the bizarre high-frequency
performance characteristic that Scott described. The high frequency of
a bias signal simply ensures that the domains are always kept in motion,
negating the effect of inertia at audio frequencies. -Gabe]

Q4.3 - What is the difference between Dolby A, B, C, S, and SR? How do each
of these systems work?

The Dolby A, B, C, SR, and S noise reduction (NR) systems are non-linear
level-dependent companders (compressors/expanders). They offer various
amounts of noise reduction.

The band-splitting system used with Dolby A NR is a relatively costly
technique, although it can deal with noise at all frequencies. The
single sliding band techniques used in Dolby B and C systems are less
costly, making them more suitable for consumer tape recording
applications where the dominant noise contribution occurs at high

The amount of boost during the compansion
depends on the signal level and its spectral content. For a tone at
-40dB at 3 kHz, the boost applied to signals with frequencies above this
would probably be the full 10dB allowed by the system. If the same tone
were at a level of -20dB, then the boost would be less, maybe about 5dB.
If the tone was at 0dB, then no boost would be supplied, as tape
saturation would be increased (beyond it's normal amount).

The single band of compansion utilized with Dolby B NR reaches
sufficiently low in frequency to provide useful noise reduction when no
signal is present. Its width changes dynamically in response to the
spectral content of music signals. As an example, when used with a solo
drum note the companding system will slide up in frequency so that the
low frequency content of the drum will be passed through at its full
level. On replay, the playback of the bass drum is allowed to pass
through without modification to its level, while the expander lowers the
volume at high frequencies above those of the bass drum, thus providing
a reduction in tape hiss where there is no musical signal. If a guitar
is now added to the music signal, the companding band slides further up
in frequency allowing the bass drum and guitar signals through without
any compansion, while still producing a worthwhile noise reduction
effect at frequencies above those of the guitar.

The Dolby B NR system is designed to start taking effect from 300Hz, and
its action increases until it reaches a maximum of 10dB upwards of 4kHz.
Dolby C improves on this by taking effect from 100Hz and providing about
15dB of NR at 400Hz, increasing to a maximum of 20dB in the critical
hiss region from 2kHz to 10kHz. Dolby C also includes spectral skewing
networks which introduce a rolloff above 10kHz prior to the compander
when in encoding mode. This helps to reduce compander errors caused by
unpredictable cassette response above 10kHz, and an inverse boost is
added after the expander to compensate. Although this reduces the noise
reduction effect above 10kHz, the ear's sensitivity to noise in that
region is diminished, and the improved encode/decode tracking provides
important improvements in overall system performance. An anti-saturation
shelving network, beginning at about 2kHz, also acts on the high
frequencies but it only affects the high-level signals that would cause
tape saturation. A complementary network is provided in the decode chain
to provide overall flat response.

When the tape is played back, the inverse of the above process takes
place. For an accurate decoding to occur, it is necessary that playback
takes place with no offsets in levels between record and replay. i.e. If
a 400 Hz tone is recorded at 0dB (or -20dB), then it must play back at
0dB (or -20dB). This will help ensure correct Dolby "tracking".

Just think about it: if a -40dB tone at 8kHz was recorded with
Dolby B on, then it would actually have a level of -30dB on tape.
The same tone, if it were at a -20dB level, would have a level of
about -15dB on tape. If the sensitivity of the tape was such
that anything recorded at 0dB actually went on tape as -10dB,
then you can see that the Dolby encoded tones would actually be
at a lower level, and the system would have no way of determining
this. It assumes 0dB in = 0dB out. Hence the signal would be
decoded with the incorrect amount of de-boost.

The Dolby SR and S NR systems provide slightly more NR than Dolby C at
high frequencies, 24dB vs 20dB, but they also achieve a 10dB NR effect
at low frequencies below 200Hz as well. This is obtained using a
two-band approach, the low-frequencies being handled by a fixed-band
processor, while the high frequencies are tackled by a sliding band
processor. This reduces the potential for problems such as "noise
pumping", caused by high-level low frequency transient signals (bass
notes from drums, double basses, organs), raising the sound level in a
cyclic fashion. Dolby SR and S also contain the spectral skewing and
anti-saturation circuits for high-level high-frequency signals that are
implemented with Dolby C. The performance of the sliding band is
improved over that obtained with Dolby B and C NR systems by reducing
the degree of sliding that occurs in the presence of high-frequency
signals. This increases the noise reduction effect available at
frequencies below those occurring in the music signal.

An additional benefit of the Dolby S NR system for consumers is that the
manufacturers of cassette decks who are licensed to use the system must
adhere to a range of strict performance standards. These include an
extended high frequency response, tighter overall response tolerances, a
new standard ensuring head height accuracy, increased overload margin in
the electronics, lower wow and flutter, and a head azimuth standard.
These benefit users by enhancing the performance of cassette recorders
as well as helping to ensure that tapes recorded on one deck will play
back accurately on any other. [Witold Waldman - witold@aed.dsto.gov.au]

Q4.4 - What is Dolby HX-Pro?

HX-Pro is a scheme to reduce the level of the bias signal when high
frequency information is present in the recorded signal. Sufficient
high frequency information will act to bias the tape itself, and by
reducing the AC bias signal somewhat, additional signal can be applied
without saturating the tape. This is a single-ended system; it
requires no decoding on playback, because it merely permits more
signal to be recorded on the tape. In theory it is an excellent idea,
and some implementations have lived up to the promise of the method,
although some other implementations have produced unpleasant
artifacts. [Scott]

Q4.5 - How does DBX compare to Dolby?

[Anyone? Anyone? -G]

Q4.6 - How much better are external microphone preamplifiers than those
found in my portable recorder?

Going by the rule that "external is better than internal," the
external preamps are likely to sound better. Besides the issue of
electrical shielding and interaction, it is simply the case that a
designer who is spending *all* his time on a project designing only a
preamp is likely to do a better job of it than a tape machine design
team that has to worry how they're going to fit the preamp into the
box and still have enough room for the rest of the tape machine. [Gabe]

Q4.7 - What is an MRL? Where do I get one?

An MRL is a reference alignment tape from Magnetic Reference Laboratory.
These tapes, available in every conceivable tape speed, tape width,
equalization, and field strength, contain alignment tones useful in
calibrating the electronics of analog tape machines.

These tapes can be ordered from many pro audio dealers. If not, you
can contact MRL directly at:

Magnetic Reference Laboratory, Inc.
229 Polaris Avenue, Suite 4
Mountain View, CA 94043

Tel: (415) 965-8187
Fax: (415) 965-8548

Section V - Digital recording and interconnection

Q5.1 - What is sampling? What is a sampling rate?

Sampling can be (roughly) defined as the capture of a continuously
varying quantity at a precisely defined instant in time. Most usually,
signals are sampled at a set of sample-points spaced regularly in
time. Note that sampling in itself implies nothing about the
representation of sample magnitude by a number. That process is called

The Nyquist theorem states that in order to faithfully capture all of
the information in a signal of one-sided bandwidth B, it must be
sampled at a rate greater than 2B. A direct corollary of this is that
if we wish to sample at a rate of 2B then we must pre-filter the
signal to a one-sided bandwidth of B, otherwise it will not be
possible to accurately reconstruct the original signal from the
samples. The frequency 2B that is the minimum sample rate to retain
all of the signal information is called the Nyquist frequency.

The spectrum of the sampled signal is the same as the spectrum of the
continuous signal except that copies (known as aliases) of the
original now appear centred on all integer multiples of the sample
rate. As an example, if a signal of 20 kHz bandwidth is sampled at 50
kHz then alias spectra appear from 30 - 70 kHz, 80 - 120 kHz, and so
on. It is because the alias spectra must not overlap that a sample
rate of greater than 2B is required. In digital audio we are
concerned with the base-band - that is to say the signal components
which extend from 0 to B. Therefore, to sample at the standard digital
audio rate of 44.1 kHz requires the input signal to be band-limited to
the range 0 Hz to 22.05 kHz. [Chris]

Q5.2 - What is oversampling?

To take distortionless samples at 44.1kHz requires that the analogue
signal be bandlimited to 22.05kHz. Since the audio band is reckoned to
extend to 20kHz we require an analogue filter that cuts off very
sharply between 20kHz and 22kHz to accomplish this. This is expensive,
and suffers from all the ailments associated with analogue

Oversampling is a technique whereby some of this filtering may be done
(relatively cheaply and easily) in the digital domain. By sampling at
a high rate (for example 4 times 44.1kHz, or 176.4kHz) the analogue
filter can have a much lower slope since its transition band is now
20kHz to 88kHz (ie half of 176kHz). The samples are then passed
through a digital filter with a sharp cutoff at 20kHz, after which
three of every four are discarded, resulting in the sample stream at
44.1kHz that we require. [Chris]

Q5.3 - What is the difference between a "1 bit" and a "multibit" converter?
What is MASH? What is Delta/Sigma? Should I really care?

Audio data is stored on CD as 16-bit words. It is the job of the
digital to analogue converter (DAC) to convert these numbers to a
varying voltage. Many DAC chips do this by storing electric charge in
capacitors (like water in buckets) and selectively emptying these
buckets to the analogue ouput, thereby adding their contents. Others
sum the outputs of current or voltage sources, but the operating
principles are otherwise similar.

A multi-bit converter has sixteen buckets corresponding to the sixteen
bits of the input word, and sized 1, 2, 4, 8 ... 32768 charge units.
Each word (ie sample) decoded from the disc is passed directly to the
DAC, and those buckets corresponding to 1's in the input word are
emptied to the output.

To perform well the bucket sizes have to be accurate to within +/-
half a charge unit; for the larger buckets this represents a tolerance
tighter than 0.01%, which is difficult. Furthermore the image
spectrum from 24kHz to 64kHz must be filtered out, requiring a
complicated, expensive filter.

Alternatively, by using some digital signal processing, the stream of
16-bit words at 44.1kHz can be transformed to a stream of shorter
words at a higher rate. The two data streams represent the same signal
in the audio band, but the new data stream has a lot of extra noise in
it resulting from the wordlength reduction. This extra noise is made
to appear mostly above 20kHz through the use of noise-shaping, and the
oversampling ensures that the first image spectrum occurs at a much
higher frequency than in the multi-bit case.

This new data stream is now converted to an analogue voltage by a DAC
of short word length; subsequently, most of the noise above 20kHz can
be filtered out by a simple analogue filter without affecting the
audio signal.

Typical configurations use 1-bit words at 11.3MHz (256 times over-
sampled), and 4-bit words at 2.8MHz (64 times oversampled). The
former requires one bucket of arbitrary size (very simple); it is the
basis of the Philips Bitstream range of converters. The latter
requires four buckets of sizes 1, 2, 4 and 8 charge units, but the
tolerance on these is relaxed to about 5%.

MASH and other PWM systems are similar to Bitstream, but they vary the
pulse width at the output of the digital signal processor. This can be
likened to using a single bucket but with the provision to part fill
it. For example, MASH allows the bucket to be filled to eleven
different depths (this is where they get 3.5 bits from, as 2^(3.5) is
approximately eleven).

Lastly it is important to note that these are all simply different
ways of performing the same function. It is easy to make a lousy CD
player based around any of these technologies; it is rather more
difficult to make an excellent one, regardless of the DAC technology
employed. Each of the conversion methods has its advantages and
disadvantages, and as ever it is the job of the engineer to balance a
multitude of parameters to design a product that represents value for
money to the consumer. [Chris]

Q5.4 - On an analog recorder, I was always taught to make sure the signal
averages around 0 VU. But on my new DAT machine, 0 is all the way at
the top of the scale. What's going on here?

Analog recorders are operated such that the signal maintains a nominal
level that strikes a good balance between signal-to-noise ratio and
headroom. Further, since analog distorts very gently, you often can
exceed your headroom in little bits and not really notice it.

Digital is not nearly as forgiving. Since digital represents audio as
numerical values, higher levels will eventually force you to run out
of numbers. As a result, there is an absolute ceiling as to how hot
you can record. If you record analog and have a nominal 12 dB of
headroom, you'll probably be okay if you have one 15 dB transient that
lasts for 1/10th of a second. The record amps _might_ overload, the
tape _might_ saturate, but you'll probably be fine. In a digital
system, those same 3 dB of overshoot would cause you to clip hard. It
would not be subtle or forgiving. You would hear a definite snap as
you ran out of room and chopped the top of your waveform off.

The reality is that digital has NO HEADROOM, because there is no
margin for overshoot. You simply must make sure that the entire
dynamic range of the signal fits within the limits of the dynamic
range of your recorder, without exception. The only meaningful
absolute on a digital recorder, therefore, is the point at which you
will go into overload. The result is the metering system we now have.
0 dB represents digital ceiling, or full-scale. The negative numbers
on the scale represents your current level relative to the ceiling.

Thus, to return to our example, if you have a transient with 15dB of
overshoot past your nominal level, you must then place your nominal
level at a maximum of -15 dB. 0 dB on the meters is the absolute limit
of what you can record. [Gabe]

Q5.5 - Why doesn't MiniDisc or Digital Compact Cassette sound as good as DAT
or CD? After all, they're both digital.

Both MD and DCC use lossy compression algorithms (called ATRAC and
PASC respectively); crudely, this means that the numbers coming out of
the machine are not the same as those that went in. The algorithms use
complex models of the way the ear works to discard the information
that it thinks would not be heard anyway.

For example, if a pin dropped simultaneously with a gunshot, it may be
reasonable to suggest that it isn't worth bothering to record the
sound of the pin! In fact it turns out that around 75 to 80 per cent
of the data for typical music can be discarded with surprisingly
little quality loss.

However, nobody denies that there is a quality loss, particularly
after a few generations of copying. This fact and others make both MD
and DCC useful only as a consumer-delivery format. They have very
little use in the studio as a recording or (heaven forbid!) mastering
format. [Chris]

Q5.6 - What is S/P-DIF? What is AES/EBU?

AES/EBU and S/P-DIF describe two similar protocols for communicating
two-channel digital audio information over a serial link. They are
slightly different in details, their basic format is almost identical,
but there are enough differences that the two are, for all intents and
purposes electrically incompatible. Both of these digital protocols
are described fully in an international standard, IEC 958, available
from the International Electrotechnical Commission.

AES/EBU (which stands for the joint Audio Engineering Society/European
Broadcasting Union standard) is the so-called "professional" protocol.
It uses standard 3-pin XLR connectors and 110-ohm balanced
differential cables for connection (no, standard microphone cables,
not even good quality cables, won't work, even though it seems they
might) and a 5 volt, differential signal.

S/P-DIF (which stands for Sony/Philips Digital InterFace, a now
obsolete standard superseded by IEC 958) is the so-called "consumer"
format. It uses what appears to be standard RCA connectors and cables,
but, in fact, require 75-ohm connectors and cables. Good quality video
"patch" cables have proven adequate (no, standard "audio" patch cords,
even excellent quality versions, have been shown not to work). The
signals are 0.5 volts unbalanced.

The actual datastream, are very similar. Each sample period a "frame"
is transmitted. Each frame consists of two "subframes", one each for
left and right channels, each subframe is 32 bits wide. In that
subframe, 4 bits are used for synchronization, then up to 24 bits are
usable for audio (the "consumer mode" format is limited to 16 bits).
The remaining four bits are used for parity (the first level of error
detection), validity, user status and channel status. 192 subframes
are collected, and the 192 user bits and 192 channel status bits are
collected into separate 24 8 bit status bytes for each channel.

The channel status bytes are interesting, because they contain the
important control information and the major differences between the
two protocol formats. One bit tells whether the data stream is
professional or consumer format. There are bits that specify
(optionally) the sample rate, deemphasis standards, channel usage, and
other information. The consumer format has several bits allocated to
copy protection and control: the SCMS bits.

Now, the notion that all of this is encoded in a standard may be
reassuring, but a standard is nothing but a voluntary statement of
common industry practice. There is a lot of incompatibility between
equipment out there caused directly by subtle differences between
interpretations and implementations. The result is that some
equipment simply refuses to talk to each other. Even THAT
possibility is stated in the standard! [Dick]

Q5.7 - What is clock jitter?

Clock jitter is a colloquialism for what engineers would readily call
time-domain distortion. Clock jitter does not actually change the
physical content of the information being transmitted, only the time
at which it is delivered. Depending on circumstance, this may or may
not affect the ultimate decoded output.

Let's look at this a little more closely. Digital audio is sent as a
set of binary digits....1's and 0's. But that is only a logical
construct. In order to transmit binary math electrically, we use
square waves. Realize that although we have two mathematical states,
we have to transmit such a construct using control voltages and

All digital audio systems start with a crystal controlled oscillator
producing a square wave signal that is used to synchronize the entire
digital audio sampling and playback processes. Now, for a clock, we
don't really care about the fact that the clock might be at state 1 or
state 0 at any given moment. That doesn't give us any information.
As a computer, I can't tell if my clock has just gotten to state 1, or
if it's been sitting there for a microsecond. Thus it isn't the
states we care about. Instead, we care about the state
*changes*....when the clock shifts from one state to the other.

Now, in a perfect square wave (no such thing exists), the change of
state would be instantaneous. BOOM...it's done. But in reality, it
doesn't work this way. Square waves contain high orders of harmonics.
Fourier teaches us that all complex waveforms are made up of simpler
waveforms. Thus, as we run through noisy electronics, long cables,
inadvertent filtering circuits, we begin to lose some of our
harmonics. When this happens, our square wave begins to lose form.

The result of this is that our nice sharp corners become rounded. So
our state changes are no longer precisely at the edge anymore, because
there is no more edge. The pointy edge is now all fuzzy. It now
depends on design of the electronic comparator circuit as to when the
clock state will change, as the stage change has shifted. The clock
is, essentially, jittering.

People love to bark out "Bits is bits. A copy of a computer file
works as well as the original." Yes, this is true. But these
jittering bits can create audible distortion during the digital-to-
analog conversion, and the industry is working hard to reduce the
amount of jitter present in digital systems.

Furthermore, emerging research is suggesting that certain types of
jitter may produce digital copies with eccentricities that result in
more jittery output on playback. The jury is still out on the
specifics however. Stay tuned. [Gabe]

Q5.8 - What kind of cable AES/EBU or S/P-DIF cables should I use? How long
can I run them?

The best, quick answer is what cables you should NOT use!

Even though AES/EBU cables look like orinary microphone cables, and S/P-DIF
cables look like ordinary RCA interconnects, they are very different.

Unlike microphone and audio-frequency interconnect cables, which are
designed to handle signals in the normal audio bandwidth (let's say that
goes as high as 50 kHz or more to be safe), the cables used for digital
interconnects must handle a much wider bandwidth. At 44.1 kHz, the digital
protocols are sending data at the rate of 2.8 million bits per second,
resulting in a bandwidth (because of the biphase encoding method)
of 5.6 MHz.

This is no longer audio, but falls in the realm of bandwidths used by
video. Now, considerations such as cable impedance and termination become
very important, factors that have little or no effect below 50 kHz.

The interface requirements call for the use of 110 ohm balanced cables for
AES/EBU interconnects, and 75 ohm coaxial unbalanced interconnects for
S/P-DIF interconnects. The used of the proper cable and the proper
terminating connectors cannot be overemphasised. I can personally testify
(having, in fact, looked at the interconnections between many different
kinds of pro and consumer digital equipment) that ordinary microphone or
RCA audio interconnects DO NOT WORK. It's not that the results sound
subtly different, it's that much of the time, it the receiving equipment
is simply unable to decode the resulting output, and simply shuts down.

Fortunately, there is a ready solution for S/P-DIF cables. Any store that
sells high quality 75 ohm RCA video interconnect (or "dubbing") connectors
also sells high-quality S/P-DIF interconnects as well. They may not know
it, but they do. This is because the signal and bandpass requirements for
video and S/P-DIF cables are the same. National chains such as Radio Shack
sell such cables, and the data seems to indicate that they are good digital

For AES/EBU, there are fewer, less common solutions. Companies such as
Canare make excellent cables. Professional audio suppliers and distributors
may be good sources for such cables. If you are handy with a soldering
iron, then you can purchase 110 ohm balanced shielded cable and make your
own (which I have done quite successfully). Cables such as Alpha Twinax,
Carol Twin Coaxial, Belden 9207 twin axial, and the like, all work well for
this application. Use high-quality XLR connectors (be warned that these
cables are 0.330 inches in diameter and are a VERY tight fit in the
neoprene strin reliefs of many connectors: warming them in hot water makes
them pliable enough to work well).

As to how long these cables can be, it's hard to say. However, a couple of
general rules apply.

S/P-DIF was NEVER intended to be a long-haul hardware interconnect. The
relevant specifications talk of interconnect lengths less than 10 meters
(33 feet). In fact, many pieces of equipment cannot tolerate cables even
that long, due to the excessive capacitance and possibly induced common
mode interference.

AES/EBU is more tolerant of longer runs because it is balanced (thus more
immune to interference) and it's run at a higher signal level (5 volts
instead of 0.5 volts). The standards "allow signal transmission up to a few
hundred meters in length."

The reality is that much is highly dependent upon the actual conditions at
hand. The requirements are that the received signal fit within certain
requirements of rise time/period and voltage level, the so-called "eye
diagram". In other words, regardless of what kind of cable you use, if it
can't move the voltage at the receiver far enough soon enough, it simply
isn't going to work.

Another complicating factor is that both protocols allow a degree of
multi-drop capability. This means a single transmitter can drive several
receivers (the last of which must be terminated with the proper termination
impedance). However, implementing multi-drop puts more stingent
requirements on impedance matching. [Dick]

Q5.9 - What is SCMS? How do I defeat it?

SCMS is the Serial Copy Management System, a form of copy protection
that was mandated by Federal law (the Home Recording Rights Act).
SCMS consists of a set of subcode flags that indicate to a digital
recorder whether or not the source may be copied. Under the HRRA,
consumers are permitted to make one digital generation, but no more.
Thus when, for instance, the consumer copies a CD onto DAT, the SCMS
flag is set on the copy, and no further generations can be made.

SCMS is only mandated in consumer machines. Any recorder sold
through professional channels, and which is intended for use in
professional applications, does not have to implement it.

There are several professional products, such as Digital Domain's
FCN-1 format converter, which allow manipulation of the SCMS flags.
These units exist so that professional engineers may adjust the
subcode bits of the recordings they produce. [Gabe]
Q5.10 - What is PCM-F1 format?

In the 1980s, before the DAT era, Sony produced a set of PCM adaptors
that enabled one to record digital audio using a video cassette
machine. These units had RCA audio connections for input and output,
as well as video I/O that could be sent to, and received from, the
VCR. At the time, these systems offered performance far in excess of
conventional analog recorders available in the price category.

Sony released many models, including the PCM-F1, PCM-501, PCM-601, and
PCM-701. Perhaps the most interesting is the PCM-601, which has
S/P-DIF digital I/O. These units are highly prized since they are the
only units that can be used to make digital transfers of F1 tapes to
modern hardware.

There are some engineers who insist that, despite the clunkiness of the
format by modern DAT standards, the F1 series was the best digital format
ever developed. To this day, it is not surprising to see an F1 encoder
on a classical recording session. [Gabe]

Q5.11 - How do digital recorders handle selective synchronization?

Selective Synchronization, or "sel-sync" as it is often called, is the
ability of a recorder to play and record simultaneously, allowing
synchronous recording of new material onto specific tracks without
erasing everything on tape. This technique is what makes overdubbing

On an analog recorder, audio tracks are discrete entities, and the
sync head is really just a stack of individual heads, any one of which
is capable of recording or playing back. Thus sel-sync is a
relatively simple matter of putting some heads into record and others
into repro.

In the digital world, the problem is highly complex. First, A/D and
D/A conversion involves an acquisition delay of several milliseconds.
Second, and more importantly, digital tracks are not discrete. Rather,
they are multiplexed together on a tape, along with subcode and other
non-audio information. So how can you replace one track and leave the
others untouched?

The answer is a technique called "read before write" (RBW) or "read,
modify, write" (RMW) which involves a second set of heads. The data
is read from the tape and flushed into a buffer, where it can be
modified, and ultimately written back to the tape. Thus when you
"punch in" on a digital deck, you are physically re-writing all the
tracks, not just the one you're overdubbing. You are not, however,
changing the data on any track other than the one you want to
replace. [Gabe]

Q5.12 - How can a 44.1 kHz sampling rate be enough to record all the
harmonics of music? Doesn't that mean that we chop off all the harmonics
above 20 khz? Doesn't this affect the music? After all, analog systems
don't filter out all the information above 20 kHz, do they?

This whole question is based on the premise that "analog systems don't
filter out all the information above 20 kHz." Indeed there are mixers
and power amplifiers and other electronic systems that are capable of
stunningly wide bandwidth, often exceeding 100 kHz, the same cannot be
said for the entire analog reproduction chain. The mechanical
transducers, microphones, speaker and phono cartridges seldom have
real response far exceeding 20 kHz. In fact, some of the most highly
regarded large diaphragm condensor microphones often used in very high
quality recordings seldom exceed 18 kHz bandwidth. Analog tape
recorders rarely have bandwidths as wide as 25 kHz, and LP
reproduction systems have similar limitations in reality.

So while it may be possible to send very high frequency ultrasonic
signals through parts of both analog and digital reproduction chains,
there are, in both technologies, fundamental and insurmountable limits
to the bandwidth that, in reality, lead to very similar actual
reproducible bandwidths in each.

Thus, one of the basic premises of the question is flawed. Analog
systems DO filter out information above 20 kHz. Further, the frequency
response and phase errors of even the very best well-maintained analog
reproduction systems have response errors far exceeding those of even
middle of the line digital equipment. Whether one person may find
those errors tolerable or even likeable or not is a matter or personal
preference that is beyond the scope of this or any other technical

There are a variety of anecdotal tales that are advanced to "prove"
that the ear can hear far beyond what is conventionally accepted as
the 20 kHz upper limit (an upper limit that, for the most part,
applies to young people only: modern high SPL music and noise levels
has lead to a widespread deterioration in the hearing of the adult
population at large, and especially amongst young males).

For example, there is an apocryphal story about Rupert Neve that
tells of a console channel that sounded particularly "bad". It was
later discovered that it was oscillating at some ultrasonic frequency,
like 48 kHz. Rupert Neve is rumored to have seized upon this as
"proof" that the ear can hear well beyond 20 kHz. However, there exist
an entire range of perfectly plausible mechanisms that require NO
ultrasonic acuity to detect such a problem. For example, the existence
of ANY nonlinearity in the system would result in the production of
intermodulation tones that would fall well within the 20 kHz audio
band and certainly would make it sound awful. Even the problem that
was causing the oscillation itself could lead to massive artifacts at
much lower frequencies that would completely account for the alleged
sound of the mixer in the complete absence of a 48 kHz "whistle."

Whether 20 kHz is an adequate bandwidth is a debatable subject.
However, several important facts have to be remembered. First, BOTH
analog AND digital reproduction systems suffer from roughly the same
bandwidth limiting. Second, digital systems using properly implemented
oversampling techniques have far less severe phase and frequency
response errors within the audible band. No analog storage and
reproduction system can match the phase and response linearity of a
digital system, both at low and high frequencies. Once those
demonstrable facts are acknowledged, then the discussion about
supra-20 kHz aural detectability can continue, knowing that, if it is
demonstrated to be significant, both systems are provably deficient.

Q5.13 - Yeah, well what about square waves? I've seen square wave
tests of digital systems that show a lot of ringing. Isn't that bad?

Square waves are a mathematically precisely defined signal. One of the
ways to describe a perfect square wave is as the sum an infinite series
of sine waves in a precise phase, harmonic and amplitude relationship.

Remember, we require an infinite number of terms to describe a perfect
square wave. If we limit the number of terms to, say, 10 terms, (such as
the case with a 1 kHz square wave perfectly band limited to 20 kHz),
there simply aren't enough terms to describe a perfect square wave.
What will result is a square wave with the highest harmonic imposed on
top as "ringing." In fact, this appearance indicates that the phase
and frequency response is perfect out to 20 kHz, and the bandwidth
limiting is limiting the number of terms in the series.

Well, what would a perfect analog system do with square waves? As it
turns out, if you take a high quality 15 IPS tape recorder, bias and
adjust it for the flattest possible frequency response over the widest
possible bandwidth, the result looks remarkably like that of a good
digital system for exactly the same reasons.

On the other hand, adjust the analog tape recorder for a square wave
response that has no ringing, but the fastest possible rise time. Now
listen to it: it sounds remarkably dull and muffled compared to the
input. Why? Because in order to achieve that square wave response, it's
necessary to severely roll off the high end response in order to
suppress the high-frequency components needed to achieve fastest rise
time. [Dick]

Q5.14 - How can a 16-bit word length be enough to record all the detail
in music? Doesn't that mean that the sound below -96 dB gets lost in the
noise? Since it is commonly understood that humans can perceive audio
that IS below the noise floor, aren't we losing something in digital
that we don't lose in analog?

You're correct in saying that human hearing is capable of perceiving
audio that is well below the noise floor (we won't say what kind of
noise floor just yet). The reason it can do this is through a process
the ear and brain employ called averaging.

If we look at a single sample in a digital system or an instantaneous
shapshot in an analog system, the resulting value that we measure will
consist of some part signal and some part ambiguity. Regardless of the
real value of the signal, the presence of noise in the analog system
or quantization in the digital system sets a limit on the accuracy to
which we can unambiguously know what the original signal value was. So
on an individual sample or instantaneous snapshot, there is no way
that either ear or measurement instrument can detect signals that are
buried below either the noise or the quantization level (when properly

However, if we look at (or listen to) much more than a single sample,
through the process of averaging, both instruments and the ear are
capable of detecting real signals below the noise floor. Let's look at
the simple case of a constant voltage that is 1/10th the value of the
noise floor. At the instantaneous or sample point, the noise value
overwhelms the signal completely. But, as we collect more consecutive
snapshots or samples, an interesting thing begins to happen. The noise
(or dither) is random and its long term average is, in fact, 0. But the
signal has a definite value, 1/10. Average the signal long enough, and the
average value due to the noise approaches 0, but the average value of
the signal remains constant at 1/10.

A somewhat analogous process happens with high frequency tones. In
this case the averaging effect is that of a narrow-band filter. The
spectrum of the noise (or simple dither) is broadband, but the
spectrum of the tone is very narrow band. Place a filter centered on
the tone and while we make the filter narrower and narrower, the
contribution of the noise gets less and less, but the contribution of
the signal remains the same.

Both the ear and measurement instruments are capable of averaging
and filtering, and together are capable of pulling real signals from
deep down within the noise, as long as the signals have one of two
properties: either a period that is long compared to the inherent
sampling period of the signal in a digital system or long compared to
the reciprocal of the bandwidth in an analog system, or a periodic
signal that remains periodic for a comparably long time.

Special measurement instrument were developed decades ago that were
capable of easily detecting real signals that were 60 dB below the
broadband noise floor. And these devices are equally capable of
detecting signals under similar conditions in properly dithered
digital systems as well.

How much the ear is capable of detecting is dependent upon many
conditions, such as the frequency and relative strength of the tone,
as well as individual factors such as aging, hearing damage and the

But the same rules apply to both analog systems with noise and digital
systems with decorrelated quantization noise. [Dick]

Q5.15 - Q5.14 - What's all this about 20- and 24-bit digital audio? Aren't
CDs limited to 16 bits?

Yes, CDs are limited to 16 bits, but we can use >16-bit systems to produce
16-bit CDs with higher quality than we could otherwise.

We are able to record audio with effective 20-bit resolution nowadays.
The finest A/D converter systems have THD+N values around -118 dB with
linearity extending far below even that. When it comes time to reduce
our word-length to 16 bits, we can use any one of a variety of noise
shaping curves, the job of which is to mix with our 24-bit audio, shift
the dither spectrum of the noise into areas where our ears are less
sensitive, thus enabling the noise component to comprise audio information
at the spectral areas where our ears are most sensitive. See Lipschitz's
seminal papers for fuller detail on this subject.

Furthermore, we often perform DSP calculations on our audio, and to that
end it is worthwhile to carry out the arithmetic with as much precision
as we can in order to avoid rounding errors. Most digital mixers carry
their math out to 24-bit precision at the I/O, with significantly longer
word lengths internally. As a result, two 16-bit signals mixed together
can produce a valid 24-bit output word. For that matter, a 16-bit signal
subjected to a level change can produce a 24-bit output if desired (except,
of course, for a level change that is a multiple of 6 dB, as that's just
a shift left or right).

The number of noise shaping curves available today is staggering. Sony
SBM, Weiss, Meridian 618, Sonic TBM, Apogee UV-22, Prism SNS, Lexicon
PONS, Waves, and, of course, the classic Lipschitz curve are just a few
of the multitudinous options that now exist. [Gabe]

Section VI - Digital editing and mastering

Q6.1 - What is a digital audio workstation?

A digital audio workstation (DAW) is one of our newest audio buzzwords,
and applies to nearly any computer system that is meant to handle or
process digital audio in some way. For the most part however, the
term refers to computer-based nonlinear editing systems. These systems
can comprise a $500 board that gets thrown into a PC, or can refer to
a $150,000 dedicated digital mastering desk. [Gabe]

Q6.2 - How is digital editing different from analog editing?

In the days of analog editing, one edited with a razor blade and a
diagonal splicing block. Making a cut meant scrubbing the tape over
the head, marking it with a grease pencil, cutting, and then taping
the whole thing back together. Analog editing (particularly on music)
was as much art as it was craft, and good music editors were worth
their weight in gold.

In many circles, analog editing has gone the way of the Edsel,
replaced by digital workstation editing. For complex tasks, DAW-based
editing offers remarkable speed, the ability to tweak an edit after
you make it, a plethora of crossfade parameters that can be optimized
for the edit being made, and most importantly, the ability to undo
mistakes with a keystroke. Nearly all commercial releases are being
edited digitally nowadays. Since satisfactory editing systems can
be had for around $1,000, even home recordists are catching onto the
advantages. More elaborate systems can cost tens of thousands of

There are certain areas where analog editing still predominates,
however. Radio is sometimes cited as an example, though this has begun
to change thanks to products like the Orban DSE 7000. The needs of
radio production are often quite different from those of music editors,
and a number of products (the Orban being a fine example) have sprung
up to fill the niche. Nonetheless, in spite of the rapid growth of
DAWs in the radio market, razor blades are still found in daily use
in radio stations. [Gabe]

Q6.3 - What is mastering?

Mastering is a multifaceted term that is often misunderstood. Back in
the days of vinyl records, mastering involved the actual cutting of
the master that would be used for pressing. This often involved a
variety of sonic adjustments so that the mixed tape would ultimately
be properly rendered on vinyl.

The age of the CD has changed the meaning of the term quite a bit.
There are now two elements often called mastering. The first is the
eminently straightforward process of preparing a master for pressing.
As most mixdowns now occur on DAT, this often involves the relatively
simple tasks of generating the PQ subcode necessary for CD replication.
PQ subcode is the data stream that contains information such as the
number of tracks on a disc, the location of the start points of each
track, the clock display information, and the like. This information
is created during mastering and prepared as a PQ data burst which the
pressing plant uses to make the glass pressing master.

Mastering's more common meaning, however, is the art of making a
recording sound "commercial." Is is the last chance one has to get
the recording sounding the way it ought to. Tasks often done in
mastering include: adjustment of time between pieces, quality of
fade-in/out, relation of levels between tracks (such that the listener
doesn't have to go swinging the volume control all over the place),
program EQ to achieve a desired consistency, compression to make one's
disc sound LOUDER than others on the market, the list goes on.

A good mastering engineer can often take a poorly-produced recording
and make it suitable for the market. A bad one can make a good
recording sound terrible. Some recordings are so well produced,
mixed, and edited that all they need is to be given PQ subcode and
sent right out. Other recordings are made by people on ego trips, who
think they know everything about recording, and who make recordings
that are, technically speaking, wretched trash.

Good mastering professionals are acquainted with many styles of music,
and know what it is that their clients hope to achieve. They then use
their tools either lightly or severely to accomplish all the multiple
steps involved in preparing a disc for pressing. [Gabe]

Q6.4 - What is normalizing?

Normalizing means bringing a digital audio signal up in level such
that the highest peak in the recording is at full scale. As we saw in
Q5.4, 0 dB represents the highest level that our digital system can
produce. If our highest level is, for instance, -6 dB, then the
absolute signal level produced by the player will be 6 dB lower than
it could have been. Normalizing just maximizes the output so that the
signal appears louder.

Contrary to many frequently-held opinions, normalizing does NOT
improve the dynamic range of the recording in any way, since as you
bring up the signal, you also bring up the noise. The signal-to-noise
ratio is a function of the original recording level. If you have a
peak at -6 dB, that's 6 dB of dynamic range you didn't use, and when
you normalize it to 0 dB, your noise floor will rise an equivalent

Normalizing may help optimize the gain structure on playback, however.
Since the resultant signal will be hotter, you'll hear less noise from
your playback system.

But the most common reason for normalizing is to make one's recordings
sound, LOUDER, BRIGHTER, and have more PUNCH, since we all know that
louder recordings are better, right? :-) [Gabe]

Q6.5 - I have a fully edited DAT that sounds just like I want it to sound on
the CD. Is it okay to send it to the factory?

This is a highly case-specific question. Some people truly have the
experience to produce DATs on mixdown or editing that are ready to go
in every conceivable way. Often these people can send their tapes out
for pressing without the added expense of a mastering house.

However, if you do not have this sort of expertise, and if your only
reason for wanting to send it out immediately is because you know that
it is technically possible to press from what you have, then you would
be advised to let a mastering engineer listen to and work on your
material. A good mastering engineer will often turn up problems and
debatable issues that you didn't even know were there. Also, any
decent mastering house will provide you with a master on a format
significantly more robust than DAT.

DAT is a fine reference tape format, but it simply is not the sort of
thing you want to be sending to a pressing plant. [Gabe]

Q6.6 - What is PCM-1630? What is PMCD?

PCM-1630 is a modulation format designed to be recorded to 3/4"
videotape. It was, for many years, the only way one could deliver a
digital program and the ancillary PQ information to the factory for
pressing. The PCM-1630 format is still widely used for CD production.

But PCM-1630 is now certainly an obsolete system, as there are many
new formats that are superior to it in every way. One of the most
popular formats for pressing now is PMCD (Pre-Master Compact Disc).
This format, developed by Sonic Solutions, allows for CD pressing
masters to be written out to CD-Rs that can be sent to the factory
directly. These CD-Rs contain a PQ burst written into the leadout
of the discs.

Some plants have gone a step further and now accept regular CD-Rs,
Exabyte tapes, or even DATs for pressing. The danger here is that
some users may think that they can prepare their own masters without
the slightest understanding of what the technical specifications are.
For instance, users preparing their own CD-Rs must do so in one
complete pass. It is not permitted, for instance, to stop the CD-R
deck between songs, as this creates unreadable frames that will cause
the disc to be rejected at the plant. [Gabe]

Q6.7 - When preparing a tape for CD, how hot should the levels be?

Ideally you should record a digital master such that your highest
level is at 0 dB, if only to maximize the dynamic range of your
recordings. Many people like their CDs to be loud, and thus they
will normalize anyway even if they don't hit 0 dB during recording.

Some classical recordings are deliberately recorded with peaks
that are significantly lower than 0 dB. This is done in order to
prevent quiet instruments such as lutes and harpsichords from
being played too loud. If you record a quiet instrument such as
a harpsichord out to 0 dB, the listener would have to put the
volume control all the way at the bottom in order to get a
realistic level, and the inclination would be to play it at a
"normal" listening level. By dropping the mastering level, the
listener is more likely to set the playback level appropriately.


Q10.3 - How do I soundproof a room?

Despite what you may have seen in the movies or elsewhere, egg crates
on the wall don't work!

First, understand what's meant by "soundproofing". Here we mean the
means and methods to prevent sound from the outside getting in, or
sound from the inside getting out. The acoustics within the room are
another matter altogether.

There are three very important requirements for soundproofing: mass,
absorption, and isolation. Actually, there are also three others:
mass, absorption, and isolation. And to finish the job, you should
also use: mass, absorption, and isolation.

Sound is the mechanical vibration propagating through a material. The
level of the sound is directly related to the size of those
vibrations. The more massive an object is, the harder it is to move
and the smaller the amplitude of the vibration set up in it under the
influence of an external sound. That's why well-isolated rooms are
very massive rooms. A solid concrete wall will transmit much less
sound then a standard wood-framed, gypsum board wall. And a thicker
concrete wall transmits less than a thinner one: not so much because
of the distance, but mostly because it's heavier.

Secondly, sound won't be transmitted between two objects unless it's
mechanically coupled. Air is not the best coupling mechanism. But
solid objects usually are. That's why well isolated rooms are often
set on springs and rubber isolators. It's also why you may see
rooms-within rooms: The inner room is isolated from the outer, and
there may be a layer of absorptive material in the space between the
two. That's also why you'll also see two sets of doors into a
recording studio: so the sound does not couple directly through the
door (and those doors are also very heavy!).

If you are trying to isolate the sound in one room from an adjoining
room, one way is to build a second wall, not attached to the first.
This can go a long way to increasing the mechanical isolation. Try
using two sheets of drywall instead of one on each wall, and use 5/8"
drywall instead of 3/8", it's heavier.

But remember: make it heavy, and isolate it. Absorptive materials like
foam wedges or Sonex and such can only control the acoustics in the
room: they will do nothing to prevent sound from getting in or out to
begin with. [Dick]

Q10.4 - What is a near-field monitor?

A near field monitor is one that is design to be listened to in the
near field. Simple, eh?

The "near field" of a loudspeaker is area where the direct,
unreflected sound from the speaker dominates significantly over the
indirect and reflected sound, sound bouncing off walls, floors,
ceilings, the console. Monitoring in the near field can be useful
because the influence of the room on the sound is minimized.

Near field monitors have to be physically rather small, because you
essentially need a small relative sound source to listen to (imagine
sitting two feet away from an 18" woofer and a large multi- cellular
horn!). The physics of loudspeakers puts severe constraints on the
efficiency, power capabilities and low frequency response of small
boxes, so these small, near-field monitors can be inefficient and not
have the lowest octave of bass and not play ungodly loud. [Dick]

Q10.5 - What are the differences between "studio monitors" and home

It depends upon who you ask. There are speakers called "monitor"
speakers that are found almost exclusively in homes and never in

The purpose of a monitor speaker is to monitor the recording and
editing process. If you buy the concept that they are but one tool in
the process (and probably the most frequently used single tool at
that), and if you buy the concept that your tools should be flawless,
than the requirements for a monitor speaker are easy to state (but
hard to achieve): they should be the most neutral, revealing and
unbiased possible. They are the final link between your work and your
ears, and if they hide something, you'll never hear it. If they color
something, you might be tempted to uncolor it incorrectly the other

There is another camp that suggests that monitor speakers should
represent the lowest common denominator in the target audience. The
editing and mix process should be done so that the results sound good
over the worst car speaker or boom box around. While such an idea has
validity as a means of verifying that the mix will sound good over
such speakers, using them exclusively for the process invites (as has
been thoroughly demonstrated in many examples of absolutely terrible
sounding albums) the possibility of making gross mistakes that simply
can't be heard in the mixing process. [Dick]

Q10.6 - My near field monitors are affecting the colors on my video
monitor. What can I do to shield the speakers?

Despite a lot of folk lore and some very impressive sounding wisdom
here on the net and in showrooms, there is effectively nothing that
you can do to the speakers or the monitor, short of moving them away
from one another, that will solve this problem.

The problem comes from the magnetic field created by and surrounding
the magnets in the loudspeaker. It's possible to design a magnet that
has very little external field, but it can be an expensive proposition
for a manufacturer. If the magnets do have large external fields, the
only technique that works is by solving the problem at the source: the
magnet. Special cancelling magnets are used, sometimes in conjunction
with a "cup" shield directly around the magnet.

You'll hear suggestions from people about placing a sheet of iron or
steel between the speakers and the monitor. That might change the
field, but it will not eliminate it. As often as not, it will make it

You'll also here from people about shielding the speaker by lining the
enclosure with lead or copper. This method is absolutely guaranteed
to fail: lead, copper, aluminum, tin, zinc and other such materials
have NO magnetic properties at all, they will simply make the speaker
heavier and won't solve the problem at all. There is but one material
that has a shot at working: something called mu-metal, a heavy, very
expensive, material designed for magnetic shield that requires
extremely sophisticated and difficult fabrication and annealing
techniques. Its cost is far greater than buying a new set of speakers
that does not have the problem, and it may not even work if the
majority of the offending field is radiated from the front of the
speaker, which you obviously can't shield.

Try moving the speakers relative to your monitor. Often, moving them
an inch or two is enough to cure the problem or at least make it
acceptable. Sometimes, placing the speakers on their sides with the
woofers (the major offenders in most cases) farthest away from the
monitor works. [Dick]

Section XI - Industry information

Q11.1 - Is there a directory of industry resources?
Q11.2 - What are the industry periodicals?
Q11.3 - What are the industry trade organizations?
Q11.4 - Are there any conventions or trade shows that deal specifically
with professional audio?

Section XII - Miscellaneous

Q12.1 - How do I modify Radio Shack PZMs?


Q12.2 - Can I produce good demos at home?

[Who wants to do this?]

[That's a request for a writer, NOT a professional opinion!]

Q12.3 - How do I remove vocals from a song?

You probably want a device called the Thompson Vocal Eliminator made
by LT Sound in Atlanta, Georgia. The device will cancel out any
vocal that is mono and panned dead-center. The unit works by
filtering out the low frequencies, phase cancelling the rest of the
signal, and then mixing the filtered bass back in. The result is a
signal with the center, common information cancelled out. Sometimes
it works well, other times it sounds awful. [Gabe]

Section XIII - Bibliography

Q13.1 - Fundamentals of Audio Technology
Q13.2 - Studio recording techniques
Q13.3 - Live recording techniques

Q13.4 - Digital audio theory and practice

* Ken C. Pohlmann, "Principles of Digital Audio," SAMS/Prentice Hall, 1993
[Excellent introduction and explanation of all aspects of digital audio
principals and practices]

* John Watkinson, "The Art of Digital Audio," Focal Press, 1989

* Francis Rumsey and John Watkinson, "The Digital Interface Handbook,"
Focal Press, 1993
[deals with interfacing standards and protocols for both audio and video]

* IEC Standard Publication 958, "Digital Audio Interface," International
Electro-Technical Commission, 1989, 1993
[THE standard!]

* Claude E. Shannon and Warren Weaver, "The Mathematical Theory of
Communication," U. Chicago Press, 1963. [The seminal work on digital
sampling...includes the original 1948 BSTJ sampling paper]

Q13.5 - Acoustics

* Leo Beranek, "Acoustics," New York, American Institue of Physics, 1986
[The bible, but heavily mathematical, very thick and obtuse, a good book
despite the Lincoln Hall disaster.]

* F. Alton Everest, "The Master Handbook of Acoustics," Tab Books 1989
[A good entry level text on acoustics, studio and listening room design,
not heavily mathematical].

* Arthur H. Benade, "Fundamentals of Musical Acoustics," New York, Dover
Publications, 1990
[A thorough book, more on the acoustics of PRODUCING music rather than

* Any good physics text book is useful for keeping the bunk at bay.

Q13.6 - Practical recording guides


Gabe Wiener Dir., PGM Early Music Recordings
A Div. of Quintessential Sound, Inc., New York
Recording-Mastering-Restoration (212) 586-4200