Microphone Types

Condenser - How it Works

Although a complicated design, essentially two plates of metal are suspended very close to each other
and electrified. When sound waves in air push the very thin (and lightweight) pieces of metal, the distance
between the two plates vary slightly. A circuit measures this change and generates a corresponding audio
signal that matches the physical sound. Since air is moving a very thin and light mechanism, condenser
mics are much more sensitive than dynamics. Due to the electronics involved, condensers are able to
produce a stronger signal level than dynamic mics. (25.1mV of output for Audio Technica 4033a
condenser vs 2.6mV for a Shure Beta 58 dynamic)

Dynamic How it Works
Very simplistic in design: a metal coil close to a magnet. The sound waves in air push the diaphragm
which generates an electrical current (the audio signal). Perhaps the world's tiniest power plant? Air is
moving a comparatively larger and heavier mechanical structure, resulting in lower sensitivity and output
compared to condensers. (2.6mV of output for Shure Beta 58 dynamic vs. 25.1mV for Audio Technica
4033a condenser)

Condenser Dynamic

Requires Phantom Power: The circuits inside the mic No Phantom Power Required: Dynamic mics generate
require power be sent from the mixer to the mic, usually their own power when the sound wave pushes the
ODEHOHGDVDVEXWWRQRQHDFKFKDQQHORUJURXSRI diaphragm through a magnet and coil, generating
channels, or board-wide with a single switch near the electricity literally out of thin air.
power button.

Self Noise: Because of the electrical circuit inherent in
every condenser mic, some level of "self noise". (i.e.,
hiss) is present and listed on the mic spec sheet.
Cheaper condensers have higher self-noise than more
expensive ones.
Self Noise: Since there are no circuits (just a coil next to
a magnet) there is no "self noise". However, due to the
much lower output of dynamics, the mixer gain knob
must be turned higher to get the same level of signal as
a comparable condenser, introducing a mixer's own
inherent noise into the audio signal path.

Ruggedness: Fragile! Drop it hard on the floor and it Ruggedness: Very rugged (Some are run over by a
very well may be damaged. truck or dropped 4 stories & still workThere are
YouTubes of this, search: SM58 truck)

Maximum Sound Level (SPL): Cannot necessarily
handle very loud volume. Condensers have a maximum
SPL limit (listed on the spec sheet), and once reached,
the signal clips and that section of the audio is lost
forever. Good condensers have a -10 or -15db Pad
button to lower the sensitivity so it can handle higher
volumes without issue. Cheaper models cannot handle
as high volumes as more expensive models.
Maximum Sound Level (SPL): Easily handles very loud
volume. No maximum SPL. Cannot clip in the mic. A
dynamic will basically handle anything thrown at it
(putting it right against a 120dB distorted electrical guitar
amp, no problem!) It is rare to find a maximum SPL listed
RQ D G\QDPLFV VSHF VKHHW VLQFH LW LV DVVXPHG LW FDQ
handle anything louder than you can physically (and
safely) hear.

Sensitivity: Very sensitive. Picks up audio near and far, Sensitivity: Not very sensitive. Only picks up what is
even quiet, distant sounds. Will generate feedback much right in front of the mic. Much less prone to feedback.
easier since it hears everything.

Transient Response: Fast transient response. Crisp, Transient Response: Slow transient response. Blurs &
accurate sound reproduction. slurs delicate and intricate details of harmonics and
overtones found in instruments such as pianos and
acoustic guitars.

:KHQ 7KH\UH 8VHG: Orchestral pickup, vocals with :KHQ7KH\UH8VHG: Kick drum, vocals with loud stage
quiet (or well managed) stage volume, stringed volume, percussion, elec. guitar cabinets, brass
instruments, choir, piano, woodwind instruments instruments

Extended Knowledge: Ribbon Mics are technically dynamic mics, but with many condenser
characteristics. They predate condensers and are mostly obsolete now, supplanted by condensers. Still
used today for a gutsy, full-bodied sound on distorted electrical guitar cabinets.
 Polar Patterns




Cardioid
:LGH  ,JQRUHV DXGLR EHKLQG WKH PLF  0RVW FRPPRQO\ XVHG  +DV 3UR[LPLW\ (IIHFW
meaning as it gets closer to the sound source, bass response and bass sensitivity
increases. When using floor monitors with cardioids, set the floor monitor directly in front
of the mic. [Recommended as a default starting pattern]





Omnidirectional

Picks up equally in front, to the sides, and behind the microphone. Doesn't have proximity
effect (stick it right up on an instrument, and it doesn't get bassier). Used in stereo AB
spaced pairs as it mitigates phase issues. [Not recommended]





Supercardioid

Directional - Good for focusing on a source that isn't going to move and you don't want to
hear anything but that source (i.e., guitar cab, attached to a snare drum, on the bell of a
horn, a vocalist that is trained to keep the mic close to mouth). Has proximity effect.
When using floor monitors with supercardioid mics, set the monitor at a 45 degree angle
off to the side facing the performer, not directly in front of the mic.





Shotgun

Contrary to popular belief, a shotgun is actually a type of pickup pattern, and not a mic
QHFHVVDULO\$VKRWJXQPLFGRHVQWSLFNXSDXGLRDQ\IDUWKHU away than a regular mic, it
just has the most narrow and directional pickup pattern. To pickup audio at great
distances, a mic attached to a parabolic dish is required (as seen on the sidelines of
football games. Parabolic mics are also used extensively for nature recordings). Shotgun
mics are widely used in the TV/film/video industry and rarely used in live performance or
recording studio situations.

Images courtesy of http://en.wikipedia.org/wiki/Microphone#Microphone_polar_patterns






Helpful Hint: Cardioids are generally used 70% of the time and supercardioids 30%. Avoid
omnidirectionals unless an application calls for them.

1

 Making Good Recordings

These are the most common pitfalls that contribute to making a recording sound bad:

60Hz Hum
What is it: Annoying low frequency E]]]]]]]]constantly underneath all the audio.
What causes it: Ground loops or electrical interference with the audio equipment.
How to fix it: Put everything that requires power (sound mixer, recording equipment, guitar amps, keyboards)
on the same wall outlet/circuit breaker. Do not run mic cables in parallel or alongside power cords or near
fluorescent lighting or power distro equipment (circuit breakers). Keep lighting equipment off the circuit. Keep
theatrical lights at full (not dimmed or partially brightened, as dimmers under load may induce hum).

Hiss
What is it: 7VssVVVVVhigh frequency noise constantly in the audio.
What causes it: Cranking the gain, cranking inputs, cranking sensitivities all raise the noise floor. Hiss is
most frequently caused by the noise floor being too high. Not only does this create hiss, but clipping and
peaking is much more likely to occur unnecessarily when the levels are run overly hot. Hiss may alternatively
be caused by loud ventilation in the room or fans running in the background yRXDUHQWFRQVFLRXVO\DZDUHRI
until listening through a recording (brain zones them out in real life).
How to fix it: Make sure gain structure is properly calibrated and remove environmental background noise
before recording. See Tracking section for instructions on setting gain structure for recording purposes.

Distortion, Over Driven Signal, Peaking, Clipping
What is it: Disgusting, distorted sounding audio.
How to fix it: Good gain structure. Run levels lower. It is better to have the levels too low while recording
than too high. Running the levels too high introduces clipping, distortion, and hiss. Make sure condenser
mics, if closely acquiring a loud source such as percussion, have their -10dB or -15dB pad button engaged.

Narrow Bandwidth Audio
What is it: $XGLRVRXQGVORZ-ILRUVPDOORSSRVLWHRIIXOOULFKDQGKL-ILRUKLJKGHILQLWLRQ
How to fix it: Mic the source closer. Perhaps not the right mic for the job. Mastering format/media should
always be uncompressed (.WAV file). Unacceptable mastering formats: MP3, MiniDisc (both are compressed
and lead to generative losses between the tracking, editing, and distribution stages). 24-bit WAV at 44.1KHz
is recommended. 16-bit WAV at 44.1KHz is second-best recommended. See Tracking section for details.

Room Noise
What is it: Too much of the room reverb is being captured in the recording. Too spacious, distant sounding.
How to fix it: Get closer! Avoid reflections from surfaces, be it walls or the floor. Get the mics closer to the
LQVWUXPHQWV 3XW JDWHV RQ XQXVHG PLFV RU WXUQ RII XQXVHG PLFV ZKHQ WKH VRXUFH LVQW EHLQJ DFTXLUHG LH
turning off solo mics when no one is performing on them). Avoid room noise, modes, bounce backs, phasing,
and stage bleed from other instruments into too many mics (3-to-1 rule). See General Mic Placement
Guidelines

Wow & flutter, Dropouts
What is it: Interference caused by the recording device or media itself. In the analog realm, the tape drive
motors varying slightly cause a warbling of the audLR,QWKHGLJLWDOUHDOPXVXDOO\GURSRXWVZKHUHWKHKDUG
GULYH FRXOGQW NHHS XS PHGLD ZHQW EDG GULYH UDQ RXW RI VSDFH RU HYHQ EDVV FDXVHG WKH PHFKDQLFDO
recording media to skip. (i.e., pounding your hand next to laptop while hard drive is recording, bumping
computer while CD is burning)
How to fix it: Use highly-rated and quality blank media (not media that has bad reviews), make sure the
media is reformatted before each use and has enough free space to record the event 1.5x the estimated
length (as some devices freak out before reaching the end). Tape cables down running to the recorder and
make sure the recorder is not disturbed during recording by placing it out of range from being bumped or
jostled accidentally by visitors or crew.

Plosives, Vocal Pops
What is it: 7KHEDVV\38+VRXQGFDSWXUHGZKHQZRUGVWKDWVWDUWZLWK3RU%DUHVSRNHQRUVXQJ
How to fix it: 8VHDZLQGVFUHHQRUSRSSHUVWRSSHURQWKHPLFDQGDOZD\VUXQD High Pass Filter (HPF) of
~100Hz on vocal channels (usually a button on the top of each channel of the mixer by the gain knob).

2

 General Mic Placement Guidelines s

Reflections = Bad
The first important factor when placing a mic on a source is to be wary of reflections. Reflections cause
phasing issues and there is no way to repair phase issues in postthe only way to fix them is to initially
track a good signal, so keep reflections at bay! On a podium, keep the mic high above the surface and
close to lip level (or pointed up and away from the surface). On guitar cabinets, try to avoid the speaker
close to the floor and mic up off the floor to avoid reflections. On a piano, be wary of location in relation to
the piano lid. Reflections cause phasing issues and make a smooth frequency spectrum appear like saw
teeth. $ KLJK ILGHOLW\ KLJK TXDOLW\ FOHDQ VLJQDO FRPHV IURP DYRLGLQJ UHIOHFWLRQV EHFDXVH ZKHQ
reflections are avoided, phase issues are avoided.

Space Them Out

If multiple mics must be in the same vicinity or miking the
same source (a choir, single acoustic guitar with two
mics, a group of trumpet players) follow the 3-to-1 Rule:
Mics should be spaced at least 3 times wider than they
are close to the source. This is because when multiple
microphones capture the same sound, each frequency
of sound hits the various mics at different times. When
played back together, phasing issues arise and the
spectrum again appears to be a saw tooth instead of a
clean, pure sound. (See figure to right)

Get Close!
Moving a mic that is picking up a band from 10DZD\WR
 DZD\ ZLOO GRXEOH WKH RXWSXW RQ WKDW PLF! Moving it to
DZD\ (halving the new distance) will now double the
signal again, or four times as strong from where it
started. The easiest way to get the best, clean, strong,
full quality signal is to keep the mics close. Never be
afraid to mic something too close. If the sound is too
GU\RUGLVFUHWH\RXFDQDOZD\VDGGDURRPUHYHUEKDOOUHYHUEFKDPEHUreverb, or plate reverb after the
fact to back the sound away from the mic artificially. But if the mic is too far away and distant sounding to
begin with from the initial recording session, there is no way to fix this after the fact. You will not damage
the mic. A mixer will handle the signal (just turn down the gain or if the signal needs to be further reduced
past the lowest gain setting, press the pad button near the gain knob).

Mic Everything!
To get the fullest, cleanest sound, put close field mics on every instrument. For example, the drum kit
should have each drum miked, the guitar player, bass player, pianoall should have their instruments
individually miked2QO\XVHJHQHUDOSLFNXSRUURRPPLFVZKHQmiking every instrument is not feasible
or practical (such as for an orchestra). With that said, always use as few mics as possible. 'RQW
overmic. Follow the guidelines above because too many open mics creates feedback and phasing issues.

'RQW)RUJHWWKH$XGLHQFH
If we are to follow the above rule to its fullest, this means miking the audience as well. Any live
performance wilOVRXQGUHFRUGHGLIWKHEDQGVRXQGVJUHDWEXWWKHDXGLHQFHUHDFWLRQDQGHQHUJ\LVZD\
off in the distance. Such an unnatural sound disengages the listener from the recording. Having good
audience reaction makes the music sound better because it involves the listener as being a part of that
live event, not a voyeur of a historical representation of the performance.

Miking the audience can be done by taking two condenser mics in ORTF and aiming them at the
audience from the inline plane of the sound reinforcement system. ORTF mics could also be in a balcony
or hung from the ceiling in the middle or rear of the hall, but then must be time aligned in post-production
with the stage pickup mics (which also means they must be recorded as discrete tracks). A gate may be
applied to these mics so only the audience reaction triggers the mics into the mix, keeping the mix clean.

3

 Stereo Mic Techniques
There are four major industry standard techniques for recording stereo:

ORTF RECOMMENDED Natural6SDFLRXV5HDO/LIH

XY NOT RECOMMENDED 1DUURZ%DVLFDOO\0RQR
AB Spaced Pair NOT RECOMMENDED %RXQF\
Mid-Side NOT RECOMMENDED 6WUDQJH([RWLF

ORTF
Place two cardioid mics so their elements are at a distance of 17cm
apart. Aim the mics so the elements are facing out at 110 degrees
from each other. 7KHUH DUHPDQ\ LQH[SHQVLYH VWHUHREDUV DYDLODEOHIRU
purchase which allow two mics to be easily and quickly setup in this fashion
on one mic stand. ORTF creates an incredibly realistic, wide, complete
stereo sonic soundstage of the source material. This is because it is
encoding a volume difference and a timing difference between the mics.

XY
Place two mics so the elements are almost touching, but at 90 degrees from
each other. Because the elements are picking up sound from nearly the same
point in space, XY creates essentially a mono soundstage of any source it is not
close to. It takes about 18dB of signal to appear in one mic over the other for a
stereo image to present itself! This means unless this technique is inside up
against the strings of a piano, or right on a harp, or on another instrument where
huge volume swings appear in a physically wide space, the stereo image and
soundstage will be extremely narrow. Horrible for large coverage (orchestral,
band) pickup. XY sounds this way because it is only encoding a volume difference
between the two mics. No timing difference information is encoded or retained.

AB Spaced Pair
Place two omnidirectional mics 50 cm from each other, in parallel. AB spaced
pair is a cousin of ORTF. Many times AB spaced pair is deployed incorrectly, by
using cardioids instead of the requisite omnis, and spacing the mics much farther
than 50cm apart. AB spaced pair sounds quite similar to ORTF, but is not as
refined. AB spaced pair sounds this way because it is only encoding a timing
difference between the two mics. No volume difference information is encoded or
retained.

Mid-Side
Place a figure-8 mic so the elements face perpendicular
to the sound source. Then place a cardioid closely above
the figure-8 mic directly facing the sound source. The
figure-8 must then be sent to two simultaneous channels
on the mixer with one phase inverted from the other, with
one panned hard L and the other hard R. The cardioid
sent to just one channel, panned C. Set the cardioid level
first to the desired sound level. Then raise the two figure-
8 channels simultaneously until the desired width of the
stereo image is achieved. Did you get all that!? This stereo technique is definitely the most
involved to setup of the techniques. Mid-Side is used frequently in the television industry or
where mono will need to be used as much as stereo. Because of its unique wiring, audio
engineers can go back later and pull a perfect mono signal if needed, or a stereo image as well.
However, this flexibility comes at a sonic price as the stereo image it provides is not as natural as AB or ORTF.

Diagrams courtesy of http://en.wikipedia.org/wiki/Stereophonic_sound
and http://en.wikiaudio.org/MS_Mid_side_microphone_technique

Helpful Hint: For checking phasing issues with stereo pairs (or two mics miking the same thing), monitor
both mics in monoif it sounds drastically different (for instance, missing entire ranges of frequencies such as highs
RU ORZV RU LQ JHQHUDO VRXQGV VWUDQJH WKHQ SKDVLQJ LVVXHV DUH SUHVHQW DQG WKH PLFURSKRQHV QHHG WR EH
repositioned.

4

 Balanced vs. Unbalanced

CONNECTION TYPE: Balanced, Unbalanced
These are terms for how audio is sent down a cable.

All (analog) audio cables use one of two standards for sending audio through a cable: balanced or unbalanced.

Unbalanced audio cables send sound in the most basic way: The signal is sent using two conductors. One conductor
is the hot line (which carries the audio signal) and the other is ground. Any noise, interference, or hum that is induced
into the line ends up audibly in the signal at the receiving end. Noise directly affects the audio signal and the receiving
end cannot distinguish what was the original intended signal and what is unintended noise that entered the line.

Balanced audio cables take the audio signal and send it down the wire twice. The original audio signal is sent down
the hot conductor. Simultaneously the same signal, but with polarity inversed, is sent down the cold conductor. A third
conductor is used for ground. Any noise, interference, or hum that is induced into the line will affect both conductors
equally in the same way. But what really sets balanced apart from unbalanced is how the receiving end handles the
signal. The receiving equipment takes the two signals, flips the cold FRQGXFWRUVSRODULW\VRLWPDWFKHVWKHKRWVLJQDOV
polarity. Now the receiving device has two audio signals that are very unique in nature. The original audio signal
transmitted is now identical on both lines in the same polarity. However, any noise that was induced into the line is
now positive polarity on the hot conductor, but has an opposite polarity carbon copy of itself on the cold conductor.
The audio signals from the hot and cold conductors are summed by the receiving device which causes two things to
happen: The audio signal now increases in strength (because two copies of the identical original audio signal were
added together) and any noise induced in the line cancels itself out by its own opposite copy being summed together.

Balanced FRQQHFWLRQVFDQWUDYHOJUHDWGLVWDQFHV!ZLWKRXWVLJQLILFDQWLQWHUIHUHQFH

Unbalanced FRQQHFWLRQVDUHJRRGIRUaDQGDUHPRUHVXVceptible to interference.

Generally, balanced cable connectors have 3 pins and unbalanced cable connectors have 2 pins. ,QFRQQHFWRUV
WKHVHSLQVWDNHWKHIRUPRIULQJVFRQQHFWRUV with three contacts are called TRS connection (Tip Ring Sleeve).
connectors with two contacts are called TS (Tip Sleeve). Although headphone jacks have three contacts, they are
unbalanced because they use 3 pins to carry two unbalanced signals in one compact connector.

BALANCED UNBALANCED

T56 XLR 76 RCA Stereo 3.5mm

+HDGSKRQH-DFN

IMPORTANT NOTE: An adapter cannot be used to connect an unbalanced signal to a balanced signal (or
vice versa)! In the chart above, adapters or converter cables can only be used to convert items among the left half
of the chart, or the right half of the chart, but an adapter cannot be used to convert something from one half to the
RWKHUGRQRWFURVVWKHFKDUWVYHUWLFDOOLQHZLWKDQDGDSWHU7RGRWKLVFRQYHUVLRQDSLece of electronic equipment is
required known as a DI box. Examples of results from interconnecting balanced and unbalanced equipment directly
without a DI box include weak signal and/or missing entire pieces of spectrum (lacking bass, lacking treble).

CLASSIC BAD EXAMPLE #1: Headphone, iPod, or Computer Output into XLR mic mixer input using an adapter
$PPVWHUHRKHDGSKRQHMDFNVKRXOGRQO\EHFRQYHUWHGWRGXDO5&$RUGXDO76EHFDXVHDKHDGSKRQHMDFNLV
2 unbalanced signals (left, right), it can only be adapted to another type of 2 unbalanced connectors.) Going from a
headphone jack to XLR is trying to connect an unbalanced output into a balanced input, which requires the use of a
DI box.

CLASSIC BAD EXAMPLE #2 Plugging an unbalanced TS connector into equipment expecting a balanced TRS
connector. Because 76DQG756 female ports look identical, it is up to you to read the manual and understand
what connection type the port is expecting. 0RVWSURIHVVLRQDODXGLRJHDUQH[WWRDQ\IHPDOHMDFN, will be labeled
81%$/$1&(' RU %$/$1&(' RU ERWK  0DQ\ VRXQG PL[HU  /LQH ,Q SRUWV ZLOO DXWRPDWLFDOO\ DFFHSW ERWK
unbalanced and balanced connection types, but confirmation is usually only found in the manual and not labeled on
the mixer itself due to space constraints.

CLASSIC BAD EXAMPLE #2.5: 6RPHPL[HUVKDYHXQEDODQFHG76RXWSXWVHLWKHUH[FOXVLYHO\RULQDGGLWLRQ

WR EDODQFHG ;/5 RXWSXWV  &RQQHFWLQJ WKLV XQEDODQFHG 76 PL[HU RXWSXW LQWR DQ DPSOLILHUV EDODQFHG 756 LQSXW
violates the rulHRIDWWHPSWLQJWRFRQQHFWDQXQEDODQFHGGHYLFHWRDEDODQFHGGHYLFH(YHQWKRXJKERWKDUHMDFNV
and physically connect, it is wrong to do so.

5

 Signal Levels

SIGNAL LEVELS: Hi-Z, Low-Z, +4dBu, -10 dBV, Line Level, Mic Level, Speaker
These are terms for how strongly audio is sent down a cable.

Signal Level Common Uses / Devices

Hi-Z Instrument pickup output used for Acoustic Guitar Pickup, Upright Bass
Pickup, Bass Guitar, Violin w/ transducer6LJQDOLVYHU\ZHDNGRHVQW
travel very far. It will basically reach from the instrument to a nearby DI
ER[QRORQJHUWKDQDZD\
Mic Level, Low-Z Microphones are low impedance sources and can be referred to as a
Low-Z or Mic Level.
-10 dBV Headphone jacks on computers, iPods, CD player red + white RCA
&RQVXPHU/LQH/HYHO outputs, DVD player audio output/LQH,QUHFRUGLQJMDFNon
computers
+4 dBu Sound mixer main outputs, Line In on sound mixers, professional rack-
3URIHVVLRQDO/LQH/HYHO mount audio equipment inputs/outputs are usually labeled as being +4 dBu
and many will have a +4 dBu / -10 dBV switch to select between the two
signal level standards.
Speaker Level, Amp Out The output of an amplifier used to drive speakers

An audio cable will use one of these standards to send signal down the line. Where problems creep in, is that some
of these standards can be wired (incorrectly) physically together, and will produce a result, but since these standards
are not meant to be used interchangeably, the audio sounds somewhere between bad (at worst) and suboptimal (at
EHVW6DGO\ZKHQWURXEOHVKRRWLQJ\RXZRQ
WNQRZZK\WKLQJVVRXQGZURQJEDGRUSRRUEHFDXVHHYHU\WKLQJ
DSSHDUVWREHZLUHGXSULJKWVLQFHLWVDOOSK\VLFDOO\FRQQHFWHG([DPSOHVRIUHVXOts from incorrectly mixing signal
levels standards include very quiet levels of audio, more hiss, and/or missing entire pieces of spectrum (lacking bass,
lacking treble).

76 756 XLR RCA Speakon 3.5mm

Signal Levels & the (Headphone)

Connectors Used
Hi-Z
Mic Level, Low-Z
-10 dbV (Consumer Line Level)
+4 dBu (Professional Line Level)
Speaker Level, Amp Out

The chart above illustrates a single connector may be carrying any one of several different signal standards. $76
connector is used with any of three possible signal levels; an XLR is used for both mic level and professional line
level interconnects (two very different signal types but the same connector is used for both). While the same
connector is used (meaning in theory a mic level and pro line level could be physically connected to each other),
these signal levels are not compatible and should not be interconnected! You can only safely use just an adapter or
converter cable when switching connector types from the same horizontal line in the above chart (meaning the signal
level remains the same between equipment, just the connector type changes). Using an adapter or cable that jumps
from one horizontal line to another, should never be used as it will introduce problems. In cases where jumping from
one horizontal line to another is required, a piece of electronic conversion equipment known as a DI box should be
used.

Be VXUHWRUHDG\RXUHTXLSPHQWVGRFXPHQWDWLRQRUDWD minimum the labeling of the ports to determine which signal

level is used with each particular port, and only interconnect ports of various equipment with matching signal levels.

SAFE: Going from 3.5mm headphone jack to dual RCA connectors (stayed on same horizontal line)
SAFE: &RQQHFWLQJDQDPSOLILHUWRDVSHDNHUXVLQJDQDPSZLWKD76output LQWRVSHDNHUV6SHDNRQinput
UNSAFE: Going from 3.5mm headphone jack to XLR (did not stay on same horizontal line)
UNSAFE: Plugging an acoustic guitar right into the Line In jack on the back of a mixer

6

 Tracking

Tracking is the process of actually recording the audio from the microphone onto a recording device.
Tracking is much more than simSO\KLWWLQJUHFRUG.

1st Pick the Recording Device to Use There are three options:

Computer Interface This is a device with microphone ports that connect the mic directly to a
computer without the need of a sound mixer.
Portable recorder This is a device that has microphone ports that record to a media inside
the recorder itself. No computer or mixer is required.
Rackmount recorder This is a device that requires a mixer to connect the mics to first, but will
record to media inside the recorder so no computer is required.

2nd Configure the Recorder Record only to WAV (also called PCM) format when tracking, not to MP3 (or other
formats). MP3s are not editable directly, throw away about 90% of the audio data, cause generative loss of the audio
each re-encode or save, and TXDOLW\YDULHVE\YHQGRUV03LPSOHPHQWDWLRQ+DUGGULYHDQGIODVKFDUGVL]HVDUHVR
large and so inexpensive now there is no reason not to use WAV when tracking. MP3s are fine for the final export of
the finished/mastered audio (distribution), but not for tracking (acquisition) and editing (manipulation). Set the
recorder for 24-bit 44.1 kHz recording (preferred if supported) or 16-bit 44.1 kHz recording if 24-bit is not available.
This will be in the menu of the device, set through switches on the device, or setup in software (refer to manual). Do
not use the 48 kHz setting. 44.1 kHz and 48 kHz sound the same; the video industry uses the 48 kHz standard with
camcorders, and the audio industry (particularly CDs) use 44.1 kHz. However, converting between the two
unecessarily can cause audio degregation due to dithering used in the conversion process.

Detour Understand Metering To correctly set gain, you must first understand there are two different metering
standards in audio: dBVU & dBFS.

The quickest and easiest way to determine which standard a meter displays LVWRORRNZKHUHWKHLV2QDG%98
meter it will be about of the way up, with positive numbers going above zero and negative numbers below. On a
dBFS meter, the zero is always at the top and only negative numbers fall below. (See figure below) Every
meter can have different scale markings so not all meters of the same standard will visually represent the same
levels. The dBVU meter historically originated with analog equipment, and dBFS with digital equipment. However,
nowadays the two are both used even on completely digital equipment so understanding the two exist and being able
to recognize one from the other is important to determine where the sound levels should hover during recording.

3rd Set Gain Where to set your levels

Gain is not set by ear, but by eye. Depending if the mixer, recorder, or computer is using
dBVU or dBFS, metering must first be determined so correct gain may be set to avoid
peaking (on analog equipment) or clipping (on digital equipment).

Analog equipment (analog mixers, analog reel tape recorders, cassette tapes) "over
saturates" when they peak. It's a gentle distortion, which should be avoided, but is not
completely destructive. The rule of thumb for setting gain on an analog mixer is to PFL the
channel(s) to be recorded. Watch the meter while the performer(s) play. Set the gain knob
so the average level is hovering around the "0 dB" mark on the dBVU meter, which will be
about 3/4 the way up. (See figure to right)

Digital equipment (CD-recorders, computer audio software, flash-based portable
recorders, DAT tapes) "clips" when they peak. A clip is a hard, flat-line in the signal where no audio information is
recorded. It sounds screeching, terrible and is distractingimmediately identifiable as "bad audio". It essentially
destroys that take or song, rendering it unusable. A clip cannot be recovered in post-production; it's data lost from the
initial recording session that can never be heard again, which is why it's so important to never clip during a digital
recording. Set the gain knob so the meter hovers around the -22 to -28 dB mark on a dBFS meter when tracking.
This may seem low, it may look low, but the meter is only showing the last section of *96 dB of dynamic range* (for
16-bit recordings) and a whopping *144 dB of dynamic range* for 24-bit recordings. So what appears low on the
meter is not low since the meter is basically only showing the last section of the entire dynamic range. The meter will
usually be lighting up only about 20-40% full when tracking on a dBFS meter.

But wait, you may ask, this all seems way too low. If I listen to that in my car or on my iPod, I will have to crank the
volume because it will be so low compared to music from my favorite artists. This is where the mastering process
picks up, to give the audio an appropriate level after it has been safely acquired with no clips.

7

 Mastering / Distribution

Mastering
During the "Tracking" session what is known as a raw track LV FUHDWHG. When played as-is, compared to pre-
recorded CDs, it will sound very low in volume, requiring a listener to turn up their volume to hear an adequate level.
To resolve this, we could raise the volume of the tracks. We can raise it up and up until the highest point of the WAV
file is just underneath the clipping point on the meter. This will safely maximize the volume without clipping. This can
be done through an automated process in most audio software by using a command known as "Normalizing". Many
people incorrectly assume normalizing has now mastered the track to an appropriate level for CD. On the contrary, a
normalized track only will be slightly louder than the raw track and will still be very quiet compared to other pre-
recorded audio CDs.

This is because professionally mastered CDs are using a procedure as a last stage in the mastering process called
Limiting. Imagine if we take the normalized audio signal, and kept increasing volume more and more so the majority
of the body of the signal was now at the top of the meter, and the peaks were off the chart. It would be loud, for sure,
but sound terrible from all the clipping caused by the peaks extending past the maximum range of the meter.

A limiter will take those peaks, and push any peaks that were jumping past the point of clip, to instead reside below
the maximum of the meter so audio is not clipped. A limiter acts as a friendly helper, watching any audio that was
straying off the meterVPD[LPXP, and pushing it back down so nothing clips! It should be clipping, but doesn't. So
the audio sounds fine, and is much louder than it otherwise would possibly be. It looks clipped, but the audio was
gently pushed back down below the peak threshold so it is not actually clipping.

Original Tracking Session Tracking Session, Normalized Normalized, and then Limited
The peaks are now as loud as they The normalized signal is made so
can go without clipping. loud the peaks extend off the chart,
but the limiter forces them back down
EHORZSHDNOHYHOZKHUHLWVVDIH
Distribution

M4A, WAV, AIFF, MP3, WMA, PCM, OGG, AAC, and AU are all file formats audio may be stored in. Each varies in
size used per minute of audio storage, compatibility across devices, and sound quality. The format that is the best
mix of small file size, high sound quality, and great compatibility is MP3. Export all mastered tracks to be distributed
to listeners as MP3 format at 192-kbps bitrate at 44.1 kHz Joint Stereo. M4A and AAC is mostly recognized by Apple
branded devices. WAV files are huge, much too large for download from the web. WMA is mostly recognized only
by computers and devices running Microsoft software (Windows, Zunes). Anything will play MP3 as it has been
around for over 14 years and will continue to be widely supported going forward, has very small file sizes, and at 192
kbps bitrate (and higher) is arguably indistinguishable from CD quality.


Guided Help: If you recorded a stereo track, open in Audacity 1.3. Zoom out so the entire length of the track
ILWVRQ\RXUVFUHHQE\SUHVVLQJ&75/)$SSO\DQ\HTXDOL]DWLRQ\RXPD\ZLVKWRSHUIRUPILUVW2QFHLWLV(4GWR
WDVWH OHWV PDVWHUWKHDXGLR Be sure all audio is selected on both tracks (Edit -! 6HOHFW $OO 6HOHFW (IIHFW SXOO-
GRZQPHQXDQGFOLFN1RUPDOL]H&OLFNWKH2.EXWWRQ to normalize1RZVHOHFW(IIHFWSXOO-down menu and click
+DUG/LPLWHU6OLGHWKHG%OLPLWVOLGHUleft to about -3 or -6 dB. Click OK and review what your resulting wave form
looks like on screen. Most of the high peaks should be gone (the limiter pushed them down to be more in line with
the rest of the average audio level.) If WKHUHV no noticeable change, undo (CTRL+Z) and try -9dB. Listen to the audio
WREHVXUH\RXGRQWKHDUWKHOLPLWHUSHUIRUPLQJ,ILWLVDFFHSWDEOHSHUIRUPDQRWKHU1RUPDOL]HDQG\RXUDXGLRLVQRZ
mastered for distribution or to be burned onto a CD. &OLFN)LOHDQGVHOHFW([SRUW6HOHFW6DYHDVW\SHDQG
FKRRVH03)LOHV&OLFNWKH2SWLRQVEXWWRQDQGFKRRVH&RQVWDQWNESV-RLQW6WHUHR&OLFN2.WKHQ6DYH
enter any information (such as Artist, Date Recorded, Song Title, etc.) and finally push OK. Your MP3 file is now
saved.

8

 Misc. Final Tips

'RQWEHDIUDLGWRXVHcolor FRGHGFDEOHVHVSHFLDOO\LIWKH\ZRQWEHVHHQE\DXGLHQFHRURQO\LQDVWXGLRWRPDNH

wiring easier and quicker.

Using a drum shield helps prevent drum set from bleeding into all the other mics on stage.

Do not buy used Shure SM58, Beta 58, SM57 microphones. Savings will only be about $10 but 80% chance it's
counterfeit.

Buy quality cables

- Cables are future-proof, so spend a little more now. TKH\UHDOZD\V
going to work with any mLFVRUWHFKQRORJ\XSJUDGHVODWHUVRLWVZRUWK
paying a bit more to get a lot longer lasting/quality cable.
- Saving a little money now results in spending more time troubleshooting
static, hum, unknurling cable over the life of the cable
- Two largest namHVLQQDPHEUDQGFDEOHVDUH&DQDUHDQG0RJDPL
- /DUJHQDPHLQYDOXHEUDQGKLJKTXDOLW\EXWFKHDSFDEOHLV
Audiopile.net
- RECOMMENDED: Canare Starquad - It has redundancy built in. Each
pin is sent using two independent wires so if one breaks or is cut down
the line, the other still carries the signal (See picture to right)
- Only time I've seem them fail is when a student rips one in half, cut
going under a door, etc. but not on its own accord. Cheap cables
constantly break under their own design.
- Get Neutrik brand connectors on each end
- ~$1/foot available at proaudiola.com
- An affordable alternative is audiopile.net quad cables, but they don't wind up! (retain kinks in the line)

Inverse Square Law

When the distance from a sound source doubles, the sound level decreases by 6dB. This is useful for
knowing how far apart to space instruments, and floor monitor placement. For instance, if the sound of a floor
PRQLWRUQHDUDPXVLFLDQLVG%DWWKHQLWZLOOEHG%DWDQGG%DWDQGG%DW This also works in
UHYHUVHPHDQLQJDG%KRWWHUVLJQDOLVREWDLQHGVLPSO\E\PRYLQJDPLFURSKRQHIURPDZD\IURPWKHVWULQJVRID
SLDQRWR0RYLQJLWDJDLQMXVWIURPWRZLOOJLYHDQDGGLWLRQDOG%KRWWHUVLJQDOVLQFHWKHGLVWance is halved
DJDLQIRUDWRWDORIG%KRWWHUVLJQDODWIURPVWULQJVWKDQDW

&RPSDULQJWZRDXGLRVLJQDOV
It's easier to discern a slight deviation going from silence to something, than it is a slight deviation from
something to another something. Therefore, the easiest way to compare the similarity of two audio signals by ear is
polarity inverting one of the two signals and summing them. Immediately heard is only the differences between the
two. Anything that was the same goes silent and only any differences between the two signals is heard. If the two
signals were absolutely identical, then absolute silence is heard.

For instance, comparing two mics in XY configuration. Take one, invert the polarity it (usually a button on the mixer),
and set both to center (C), which effectively sums the two channels. Now talk into the mic or listen to audio through
the mic. You will hear almost nothing, since the same source is going into both at the same time. However, you will
hear some audio. What you are hearing is the difference between the two mics. It may be differences in the mic
itself, differences in the audio hitting one sooner than the other because of distance or angle, it could be differences
because one has higher gain set than the other.

7LPHDOLJQLQJIDUILHOGZLWKFORVHILHOGPLFV
Time aligning mics on stage picking up the band with mics halfway back in an auditorium in the middle of the
audience (for natural reverb and audience pickup) can also use the polarity invert and sum technique. This can be
done by inverting polarity of the close field mics and summing with the far field mics. Adjust the delay of the close
field mics (in milliseconds) to match the time it takes sound to reach the far field mics. Keep adjusting and when the
sound is the quietest, they are most in sync. Then revert the polarity on the close field mics to match the original
polarity and now the mics are time aligned. Rule of thumb: Sound travels 1 foot every millisecond.

Post Production
When editing audio, use studio reference monitors. Do not edit or master on headphones alonecertain
frequencies, particularly bass, are accentuated since the speakers of headphones are against your ears. And never
XVH D VHW RI FRPSXWHU VSHDNHUV WR HGLW SDUWLFXODUOy those with treble/bass adjustment knobs and those with an
external subwoofer). Reference monitors should be installed at ear height. Edit with the volumes low. Quieter is
better than louder, for rule of thumbbecause the sound will go up and sound OK, but a loud mix will not sound good
quiet (one instrument will stick out). Continually throw on a pair of headphones for stereo imaging referencing, to
make sure the stereo image is how you want it and sounds correct.

9

 Equipment Suggestions

2- Track Recording Interfaces Reference Monitors
M-Audio MobilePre Mk II
Yamaha HS50M
PC / Mac
Records using computer (USB)
'ULYHU
$120
$200 Per Speaker

PreSonus AudioBox USB
PC / Mac
Records using computer (USB) KRK Rokit 8
$150 'ULYHU
$250 Per Speaker

PreSonus FireStudio Mobile

Mac
Records using computer (FireWire)
$250

ART USB Dual Pre Multitrack Recording Interfaces
PC / Mac
Tascam US-1800
Records using computer (USB)
PC / Mac
$50
8 XLR Mic Inputs
Records using computer (USB)
$300
Tascam DR-40
SD Card
Records to internal SD Card PreSonus FireStudio Project
XLR inputs w/ phantom power Mac
$200 8 XLR Mic Inputs
Records using computer (FireWire)
$400
Tascam SS-R100
SD, CF Cards & USB Drives
PreSonus StudioLive 16.4.2
Records to removable media
PC / Mac
Requires sound mixer
16 XLR Mic Inputs
(mics cannot directly plug in)
Full featured digital mixer
$500
Records using computer
$2,000

Images courtesy of their respective manufacturer

Audio Editing Software

Audacity 1.3 Free Basic, straightforward, great starting point audio recorder / editor
PreSonus Studio One 2 Artist $100 Mid-level capability audio editing suite
Avid Pro Tools 10 for Students $300 Industry standard, extremely powerful (Normally $700)
VLC Free Universal audio format file player. Plays any audio file! (No editing)

1RWH0DQ\KDUGZDUHYHQGRUVEXQGOHD/LWHYHUVLRQRIIXOO-featured audio editing suites with their equipment. For instance, M-

Audio bundles a lite version of Pro Tools, PreSonus a lite version of Studio One. So buying a piece of hardware will help get your
feet wet with a high-end editing suite without the initial cost!

Vendors

Full Compass fullcompass.com Sells all equipment

0XVLFLDQV)ULHQG musiciansfriend.com Sells all equipment
SameDayMusic samedaymusic.com Sells all equipment
B&H Photo Video bhphoto.com Sells all equipment
Pro Audio LA proaudiola.com Cables
Audiopile audiopile.net Cables & Adaptors

10


ORTF CHART

11
Place chart over top of microphones. They should be this distance
apart (17cm) and angled out at this angle (110). The tops of the
microphone should be directly underneath the black lines, with the
microphone diaphragm facing out following the direction of the red
arrows.