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Archive-name: AudioFAQ/pro-audio-faq
Last-modified: 1996/06/25
Version: 0.9

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

	  Edited and compiled by Gabe Wiener (gabe@pgm.com)

Many thanks to all who have contributed to the FAQ.  Individual
contributions are credited at the end of their respective sections.
The core FAQ writers are listed below.  The rubric in brackets will be
used to indicate who has written a particular section.

	Scott Dorsey		kludge@netcom.com	[Scott]
	Christopher Hicks	cmh@eng.cam.ac.uk	[Chris]
	David Josephson		david@josephson.com	[David]
	Dick Pierce		dpierce@world.std.com	[Dick]
	Gabe Wiener		gabe@pgm.com		[Gabe]

 *  Each author maintains and asserts full legal copyright on his 
 *  contribution to the FAQ.  Compilation copyright (C) 1996 by 
 *  Gabe M. Wiener.  All rights reserved.  Permission is granted for 
 *  non-commercial electronic distribution of this document.  
 *  Distribution in any other form, or distribution as part of any 
 *  commercial product requires permission. Inquire to gabe@pgm.com.

---------
TABLE OF CONTENTS FOR THE FAQ:

Section I - Netiquette

 Q1.1 - What is this newsgroup for?  What topics are appropriate here, and what
        topics are best saved for another newsgroup?
 Q1.2 - Do I have to be a "professional" to post here?
 Q1.3 - I need to ask the group for help with selecting a piece of equipment.
        What information should I provide in my message?

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?  Why not just use regular voltage and power measurements?
 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.  Is it okay to send it to the factory?
 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?

--------
THE FAQ:

Section I - Netiquette

--
Q1.1 - What is this newsgroup for?  What topics are appropriate here, and what
       topics are best saved for another newsgroup?

  This newsgroup exists for the discussion of issues and topics related
  to professional audio engineering.  We generally do not discuss issues
  relating to home audio reproduction, though they do occasionally come
  up.  The rec.audio.* hierarchy of newsgroups is as follows:

  	rec.audio.pro		Issues pertaining to professional audio
	rec.audio.marketplace	Buying and trading of consumer equipment
	rec.audio.tech 		Technical discussions about consumer audio
	rec.audio.opinion	Everyone's $0.02 on consumer audio
	rec.audio.high-end	High-end consumer audio discussions
	rec.audio.misc		Everything else

  Please be sure to select the right newsgroup before posting.

--
Q1.2 - Do I have to be a "professional" to post here?

  No.  Anyone is welcome to post on rec.audio.pro so long as the messages
  you post are endemic to the group in some way.  If you are not an audio
  professional, we would ask that you read this FAQ in full before posting.
  You may find that some of your essential questions about our field are
  answered right here.  But if not, feel free to ask us.

--
Q1.3 - I need to ask the group for help with selecting a piece of equipment.
       What information should I provide in my message?

  If you are going to post a request for advice on buying equipment,
  please provide the following information.

	Your application for the equipment
	What other equipment you will be using it with
	Your budget for the equipment
	Any specific requirements the equipment should have

  There is nothing worse than messages like "Can anyone recommend a DAT
  machine for me to buy???"  Sure we can.  But what do you want to _do_ 
  with it?  We can recommend DAT machines for $400 or for $14,000.

=====
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
        necessary?

  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
  becomes:

	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
  power.

  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
  instances.

  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.
  [David]

--
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
  conductor path and couple to each other as a result.

  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
  screening.

  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

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