Recording Handbook - Hol.gr

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Από το Web Site του Αντώνη
http://users.hol.gr/~antony07
BEGGINER’S
RECORDING
HANDBOOK
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Table Of Contents
1. Sonic Fundamentals
a. Loud vs. Soft
b. Highs vs. Lows
c. Speakers and Amplifiers and Rooms
2. Getting Your Sound On Tape
a. The Analog Recording Process
b. The Digital Recording Process
c. Theory of Multi-track recording
* synchronization of tape machines
3. The Sound Source
a. Voices, Horns and Acoustic Piano
* dynamic vs. condenser microphones
* ribbon, PZM and other special types
* signal processing
b. Guitars, Basses and Things With Strings
* amplifiers vs. direct
* preamps and FX devices
c. Samplers, Synthesizers and Drum Machines
d. Real Drums and Real Drummers
4. The Mixing Console
a. Inputs and Outputs
* in-line vs. split consoles
* input trim and fader output
* what does -10 and +4 mean?
b. Equalization
* graphic, parametric, tube EQ's etc.
* vintage vs. new technology
c. Effects Sends and Returns
d. Insert Points and Patchbays
5. Mixer Automation
a. SMPTE
b. VCA vs. Moving Faders
c. MIDI automation
6. Effects Devices
a. Reverb
* Halls , Plates and Rooms
* Reverse Reverbs and other Perversions
* "Backwards" Reverb
b. Echoes and Delays
c. Flanging, Chorus and Phasing
d. Harmonizers and Exciters
7. The MIDI Revolution
a. Sequencers
* Hardware and Software Based
* MIDI Clock, MIDI Timecode(MTC) and MIDI Machine Control(MMC)
b. Synthesizers
c. Samplers
d. Computers
* digital editing
8. Recording With Your PC
a. Software
* What do you want to do?
* Stereo? Multi-track?
* Ins and Outs
* CD-r software
b. Hardware
* The Computer
* RAM
* Monitors
* PCI/ISA slots
* Power Supply
* Hard Drives: SCSI vs. IDE, USB and FireWire
* Removeable Drives
* CD-r's
* The Soundcard
* Wordclock
* Drivers
* Cost?
By Antony’s PC
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1. Sonic Fundamentals
a. Loud vs. Soft
Sound is moving air. Our ears are designed to be sensitive to these vibrations and interpret them. In music,
the term, "Dynamics" refers to whether a sound is "soft" or "loud". The ability of a recording medium to
reproduce the difference between soft and loud is called its "Dynamic Range". Vinyl records and cassette
tapes have a limited dynamic range of 20 db or so while modern CD's and Digital Audio Tape(DAT) are
capable of full dynamic range; that's 100 db! The limiting factor of how much of that range you get to
actually hear is determined by the speakers, and amplifiers and the room you're listening in. Read on...
b. Highs vs. Lows
We've all heard terms like "bright", "dull", "deep" and "thin" used to describe music. Two major factors
complicate this affair. The first is that we all hear the same thing differently; one person's "bright" is
another person's "dull". The second is the accuracy or lack thereof, of our sound source, i.e. the speakers
and amplifiers. Technically, the audible frequency range for human hearing is 20 Hertz(Hz) on the low
end and 20 Kilohertz(Khz) on the high end. Most people's hearing range falls between 40Hz and 16 Khz
and in fact, the specified frequency range of FM radio is 50Hz to 15Khz.
A typical car radio, boom box or home stereo has two EQ knobs on it. The "Low" and "High" knobs are
usually centered at 100 Hz and 10 Khz respectively with a broad "fixed Q". "Q" refers to the range of
frequencies affected by the boost or cut and is expressed in octaves. Their effect is not subtle but for
consumer applications this is simple, convenient and usually sufficient. The loudness button is simply a
low frequency boost that compensates for the apparent lack of low frequencies at low listening levels.
c. Speakers and Amplifiers and Rooms
This is the last step before your ears get to do their thing. Any problems here affect the sound reproduced,
and thusly, your ability to interpret what you hear. The amp, speakers, and the room they are in, all make
up the listening enviornment. When your mix sounds great in the studio and terrible everywhere else, you
know something is wrong.
"Flat" is a term used to describe a system that reproduces all frequencies, equally, more or less. Some
people spend thousands trying to achieve a "flat" room. As for me, it's great on paper, but it's not always
great for music! As long as I know what the speakers and room are doing, I can deal with it. I like to mix
on near-field systems at moderate SPL levels. This tends to decrease the affects the room may have. My
current favorites are the Genelec 1031 A's, a compact 2-way self-powered monitor. They don't lie to me.
Alesis and the Event 20/20 are recent entries into the more inexpensive 2-way self-powered monitor
sweepstakes.
Listening level is a very subjective matter, but the ear does respond to frequencies differently at different
volumes. Constant loud levels tend to dull the high end response of the ear, while at low levels the low
frequencies are not as apparent. As with other things in life, drugs and alcohol also affect the ears, and
usually not in a good way. BACK TO INDEX
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2.Getting Your Sound On Tape
Until recently, "Analog" was the only kind of recording available to most musicians. The wide availability
of DAT recorders, Alesis ADAT 8-tracks, Tascam DA-88 8-tracks and hard disc recorders like the Emu
Darwin, Akai and Vestax machines have forever changed that situation. Still, the process is the same even
though there are different technical considerations and format specific issues to address.
a. The Analog Recording Process
Analog recording devices use a plastic tape coated with magnetic particles moving across a magnetic
recording head at a constant speed to record and playback. There is always an "erase head", first in the
tape path, to erase and re-align the tape particles before they hit the "record head". In the "two-head"
machine there is one head for both recording and playback. The "three-head" design features one head
dedicated to recording, the "sync head", and another for playback, "the repro head". Professional machines
have three heads.
There is a limit to the intensity of the signal that the tape particles can actually absorb and reproduce. The
two parameters that interact to maximize the tape's ability to correctly record and playback are "tape
speed" and "bias". At a faster speed, there is more tape area for a given signal, i.e. more tape particles to
record. Most professional analog multitrack recorders run at 30 ips (inches per second). "Bias" is a process
that was discovered by accident. It was found that when a high frequency signal, 100 Khz or so, much
higher than human hearing, was recorded along with the normal signal, the magnetic particles did a better
job of recreating the higher frequencies.
It is a complicated process and there are lots of things to go wrong! The tape machine must be
mechanically and electronically aligned to very fine specifications. First, to ensure that it physically
handles the tape gently during shuttling, rewind and fast forward. Although tape formulations have
improved greatly over the years, mechanical problems can damage the tape by stretching or wrinkling it.
There is no error correction for this! Treat your tapes with care and respect. Other problems include loss of
particles off the tape, called shedding, speed fluctuations which produce "wow and flutter" and improper
tape to head contact.
Furthermore, the electronics have to record the input signal and play it back faithfully. This is where tones
on your master tapes becomes so important. They are required to properly align the electronics in the tape
machine so when you work at different studios, your tape sounds like you remembered. When all these
parameters are aligned correctly, you stand a good chance of hearing back a reasonable facsimile of what
you recorded previously.
b. The Digital Recording Process
The digital recording process is far simpler mechanically, but much more involved electronically. The
input signal is sampled 1000's of times per second and each acoustic slice is given it's own digital number,
consisting of 0's and 1's. Theoretically, the "analog-to-digital converter" (ADC) recieves the analog input
and converts it into a stream of numbers and conversely, the "digital-to-analog converter" (DAC) reverses
the process.
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The "sampling rate", or how many times per second the sound is sliced is the main factor in how well the
sound will survive its digitization. CD's are sampled at 44.1 K or 44,100 times per second, and that has
become an industry standard. Some formats offer 48 K sampling as well. DAC's and ADC's aren't created
equally however and there are differences in how these machines sound, despite the theoretical
consistency of 0's and 1's!
Digital tape machines use mechanical transports and plastic tape as a storage medium for the digital
information. The Alesis ADAT and Tascam DA-88 are examples of new inexpensive digital multi-tracks.
Another approach gaining acceptance are hard disc recorders. Some have computers with software as
front-end controllers, like the Digi-Design and Soundscape machines, while others are dedicated boxes
you plug hard discs into for storage, like the EMu Darwin, Vestax and Akai recorders.
With these random access digital recorders, the size of the hard discs limits the amount of recording time.
Locating is a snap, as is editing. When this approach is combined with a computer as the interface, you
have a powerful word processor for music. Anyone who has used a Mac or Windows on an IBM knows
how to drag and click with a mouse and that's basically how you manipulate the sound files.
c. Theory of Multi-track Recording
Multi-track recorders are simply tape machines that allow you to record tracks and then overdub
additional tracks in any order. For instance, you might record a drummer on four tracks, then go back and
record a guitar part, etc. To do this, the tape machine must be able to record one track while its playing
back the others. In an analog machine, it must do this from the same recording head. This is the job of the
"sync head".
Digital machines don't rely on sync heads and repro heads, they're just reorganizing 0's and 1's. Depending
on the device, sometimes the tape based digital machines are not as flexible as the random access
machines.
It is possible to "lock up" more than one multi-track tape machine to get more tracks. This is usually done
with two identical machines and SMPTE. SMPTE is an acronym for a timecode that was originally
developed for the motion picture industry. It sounds like a high pitched squeel but to devices that can
"read" it, it looks like a running clock. For lock up, we would "stripe" two multi-track tapes; for one song
we might need five minutes, so we set the SMPTE "writer" to write from 0:00:00:00 to 5:00:00:00
minutes.
SMPTE is displayed as "hours:minutes:seconds:frames:sub frames", although not all devices read
subframes (there are 80 subframes). There are four types of SMPTE. They are 30 frame per second (fps)
drop frame, 30 fps non-drop frame, 25 fps, and 24 fps. In the United States, audio professionals generally
use 30 fps non-drop frame and in England and Europe, they use 25 fps. The 30 fps drop frame, sometimes
called "29.97", is used for video and film applications in the United States.
As has become the industry practice, we record this SMPTE on the highest edge track on each tape. For
example, track 8 on an 8-track, track 24 on a 24-track, etc.. So, now we have identical "striped" tapes on
their respective multi-tracks. The next step is to use a synchronization device designed to read the SMPTE
off each machine and control the motors of both to keep them locked together. One machine becomes the
"Master" and the other, or "Slave", chases the master machine. Two of the most popular professional
systems that do this are the Lynx and Adam Smith synchronizers. This is the basic concept and it is
possible, with the right interfaces and connections, to lock up different types of multi-tracks, VCR's,
timecode equiped DAT machines, digital editors, etc.
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The concept is simple, the execution can be complicated. The most important thing to remember is what
kind of code you've got. Keep good notes! BACK TO INDEX
3. The Sound Source
There are only two ways to get your sonic information onto the tape, through a microphone or directly
from an electronic output. In general, the quality of what comes back is affected by the quality of the
equipment the signal passes through.
a. Voices, Horns and Acoustic Piano
While the human voice is the most dynamic, all of these instruments present a similar problem to the
engineer. How can we preserve the performance, that is the soft and loud of it, and get it accurately on
tape? With these instruments, we usually have to use a microphone.
The two main types of microphones are "dynamic", which have no active electronics involved in
amplifying the input signal, and "condenser", which require either batteries or "phantom power" to power
their electronics. Both types have a thin membrane, called the diaphragm, that vibrates and that physical
vibration is translated into an electronic signal. In general, condenser mikes are brighter and have a
broader frequency response, but they are more fragile. That's why you usually see an SM57, a general
purpose dynamic mike, in the lead singer's hands at a concert. They can withstand a lot of abuse.
Classic condenser microphones like the Neumann U-47 and AKG C-12 use vacuum tube electronics and
are treasured for their unique sound. They are rather large and have diaphragms 2 inches in diameter.
Ribbon microphones are another vintage design that incorporates a thin rectangular strip as it's diaphragm,
hence the name. PZM designs are a relatively new invention. They work on a completely different
principle and don't look anything like traditional microphones.
The signal created by the microphone is very small and it is the microphone pre-amp that increases this
level to what is known as "line-level" for interfacing with the mixing board. This is yet another link in the
chain with it's opportunity to affect the sound, and they do.
Everyone has their favorite microphones and pre-amps for different situations and most do color the
sound. The important thing is whether you like that color and if it's appropriate for the particular situation
at hand. Here again, we run into the concept of "flat frequency response" and again it is relatively
meaningless. Most microphones are not "flat" and some are better suited for certain jobs than others. As
always, you need a reference and in this regard, frequency response charts and the like can be useful.
Rules are made to be broken.
Signal processing is another powerful weapon for your sonic arsenal. The judicious use of compression
can be a big help in lots of situations. Compressors were originally developed to compensate for the
limited dynamic range of analog tape. Basically, they make the soft parts a little louder and the loud parts
a little softer. This performs the dual function of keeping the soft passages higher above the "noise floor"
and preventing the loud parts from getting too loud and peaking into distortion. Most compressors allow
you to change the "range" (1:1, 2:1, 4:1, etc.) and the "attack time" and "release time" of the effect.
Expanders and Limiters are related to compressors. Expanders make the soft parts softer and the loud parts
louder. Like compressors, you can set the range, and attack and release times of the effect. With limiters
you can set a threshold that cannot be exceeded. Noise Gates simply do what their name implies by
shutting off the signal path when there is no input.
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. Guitars, Basses and Things With Strings
String instruments can be recorded acoustically with microphones or directly if they have pickups. There
is a different sound to each and in different situations, one may be more appropriate than the other. Often,
both are recorded simultaneously and blended together in the mix. Electric guitars and basses are recorded
through microphones on the amplifiers and direct to be blended or used seperately later in the mix. All the
just mentioned signal processing definitely applies here, too.
The Rockman headphone guitar amp, invented by Tom Scholz in the mid 1980's, started the revolution in
small, electronic amp simulators. Rackmount guitar pre-amps have gotten very sophisticated in the last
few years, offering tube pre-amp stages, multi-effects, MIDI and memory capabilities. There's still nothing
quite like a Gibson Les Paul into a 100 watt Marshall stack, but that can get dificult for the neighbors!
c. Samplers, Synthesizers and Drum Machines
All of these devices have direct outputs and can be connected straight to your mixer. A lot of them also
include built-in effects and it's up to you whether you want to print them "wet" or "dry". It will probably
depend on how many tracks you've got to play with, but you can print sounds and their effects seperately.
This way if you change your mind about that "big reverb", you're not locked in.
Playing one sound at a time to tape has never been a problem but these days, there's a lot of sequencing
going on and most of these boxes only have 2 or 4 outputs. In 4 or 8-track recording, this is an asset and
you just have to mix the sounds from inside the box. In a professional 24 or 48-track situation, you'll want
individual sounds on individual tracks. Two ways to do this are mute the sequencer tracks and record each
sound in successive passes or turn down the internal volumes of all the sounds and turn one on at a time as
you make passes.
Drum machines are a powerful tool and have become a mainstay of modern music making. They all have
unique sounds and many have become associated with specific types of music i.e. the Roland 808 and
EMu SP-12 with rap and the Roland 909 with dance. Some also have sampling capabilities and built-in
audio triggering for replacing sounds off tape. Earlier models were designed with pads for programming,
and some people still prefer that. Now most are available as sound modules like any other synth and the
programming is simply done from the keyboard and sequencer.
Samplers are basically digital recorders. The earliest models could only record and playback one sound at
a time. In the early 80's, many a snare was replaced by hand with a steady index finger and the AMS
sampler. Two of the earliest MIDI samplers were the Akai S-612 and Ensoniq Mirage. They were 8-bit,
which refers to their sampling resolution. Technically, the higher the resolution, the more accurate the
sample. Again, specs are one thing and sound is another. These boxes all sound different and they do what
they do, differently. If you like the sound of your old 12-bit box, then go for it.
Polyphonic samplers began to appear in the mid 80's, among them the Casio FZ-1 and Akai S-900.
Resolution evolved to 12-bit and then 16-bit and MIDI made these boxes even more powerful. In the
studio, samplers are great for quick repairs, flying vocals around or moving tracks. A sequencer and a
sampler locked to tape in the mix can be real handy for those last minute repairs and arrangement changes.
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d. Real Drums and Real Drummers
There are two things that you need to get good live drum sounds, a properly tuned, great sounding kit and
a properly tuned, great drummer. If you can only get one, go with the human. You can always replace the
sounds.
The basic approach to drum miking involves a seperate mike for the kick, the snare, the hat, the toms and
one or two overheads to get the cymbals and the room sound if there is one. Another snare option is to put
one mike underneath and one on top. Some of the great drum sounds from classic rock records were
recorded with two mikes on the whole kit! The miking techniques should reflect what kind of drum sound
you're going for.
Compression can be a big help when recording drums because of the transient nature of the instruments.
Depending on the parts being played, use it as needed. Sometimes gates can be helpful as well, especially
when the rhythm section is being recorded in the same room. Noise gates with "sidechain" capability
allow you to select what frequencies will open the gate. Another excellent device for this application is the
"Kepex" expander/gate whose operation is frequency dependent.
Many engineers like to slam high levels on to analog tape to get the natural "tape compression" sound.
Some even go to the lengths of recording drums on 2" 16-track analog and then transferring that to another
format to complete the project. Hey, if you've got the budget and the time, go for it. BACK TO INDEX
4. The Mixing Console
The mixer is just that. We use it to organize our signals going to the tape machines, to organize what we
need to hear back from the tape machines, to monitor playback from our mixdown DAT, 2-track or other
stereo sources, and to add effects to whatever is needed. In short, it is the heart of the multi-track studio.
a. Inputs and Outputs
Recording consoles are designed to be connected to multi-track tape machines. They provide seperate
mixer inputs for our sound sources(mics and line inputs) and the tape returns (signal playback from the
multi-track) and multiple outputs from the mixer to the tape machines (both as "direct outs" from
individual channels and through the "bussing matrix"). In addition, the input channels with a choice of line
or microphone input also offer equalization, effects sends, pan, buss send options and a fader for volume
on each channel strip.
In-line consoles include the input section and tape return level and pan on the same physical channel strip.
Split console designs have seperate channel strips for inputs and tape returns, usually with less EQ and
effect sends on the tape returns (the "monitor section").
Semi-pro and home recording gear operates at a -10 level while professional equipment operates at a +4
level. Without getting too technical, this means you have to pay attention to the particular input and output
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levels of your boxes and how you interconnect them. When it's not right, it often hums or sounds a little
screwed up. It still might work, but it will give you much less than optimum performance.
To be sure that the sound at the source gets "through the gear and back to your ear", you need to check the
"gain stages" in the "signal path". Distortion can crop up in several places. Step one is your ear! Make sure
the sound at the source is what you want.
Step two is the microphone or direct output of the guitar, synth or whatever. In the case of a microphone,
the levels must be set carefully to ensure faithful reproduction of the input. Some mikes have a "pad"
switch on them as do most mixers, to prevent overloading the input level.
Step three is the input channel to the board. The level here must be set so that the signal doesn't overdrive
the channel electronics. Once that is right, the signal can be sent to tape through the "bussing" matrix (or
"group outs" as they are sometimes called) or the direct out on the channel. Obviously with multiple
signals to one track, they must go through the group outs and the level to tape is controlled by the group
output level.
Generally, in the analog world, very bright sounds get printed a little lower, to help prevent crosstalk bleed
to adjacent tracks. Sharp transient sounds and low frequency stuff like bass can be printed hot to take
advantage of tape compression. Slamming the tape isn't against the law, but make sure that's the sound
you want. This is only an analog phenomenon, overdriving digital recorders results in highly unusable
audio!
Tape return levels are optimized for direct connection to the machine so if this has been done correctly,
there is only one place left for distortion to be created... your monitor system. As long as you are aware of
how your speakers and listening enviornment are affecting the sound and you listen within the volume
limits of your amp and speakers, you should have a baseline for clean signal reproduction. Now, you can
introduce distorton at any of these points in the chain to any effect you prefer.
b. Equalization
"Equalization" is the term used to describe the process of changing the balance between high and low
frequencies. Equalizers allow us to selectively boost and/or cut specific frequencies or bands of
frequencies. "Q" refers to the width or range around the centered frequency that we are EQing, that is also
affected when we boost or cut. A narrow "Q" would be .2 of an octave, a wide "Q" would be 3 octaves.
There are many types of equalizers and they get used in many different ways by different people. In
general, "Parametric Equalizers" allow for very specific effect with adjustable Q and frequency control for
each frequency band. "Graphic Equalizers" feature as many as 31 individual sliders centered on fixed
frequencies. Tube equalizers utilize vacuum tubes in their circuits as oppossed to transistors ("solid state")
and are often preferred for their warm sound.
All mixers provide some kind of EQ, switchable on or off, in the signal path. These days semi-pro
consoles usually feature a couple of overlapping bands of semi-parametric EQ on the low-mids(200-2K)
and hi-mids(1.5K-7K), and one EQ each for the low(100 hz) and high(10K) bands with shelving switches
and low-frequency roll-off. Professional consoles offer fully- parametric designs and more overall
flexibility, as you might expect.
Since we can't all afford Neve VR consoles at home, another option for small studios is outboard
equalizers. Get a couple of good ones and insert them into the signal path and print through them to tape.
This will definitely take your sounds up a notch without totally blowing your college fund.
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c. Effect Sends and Returns
There are several ways to get signal to your effects and to hear those effects back. The easist is the
dedicated sends and returns on the mixing board. Sends can usually be switched between pre and post EQ.
Returns generally have little or no EQ, so if you want to EQ effects, that's one more reason to have more
channels than tape machine tracks. If you have more effects than sends, repatch the busses as sends or use
the direct outs to get into your other effects.
d. Insert Points and Patchbays
It's really nice to plug everything into the board and not have to mess with it. This is where patchbays are
a necessity and incredible convenience. Every in and out on the board, all tape machine inputs and returns
and all inputs and returns from your effects are duplicated in the patchbay. Every channel also has an
insert point as well for individually accessing the signal path. When it's all plugged in, you can change,
rearrange and repatch it all from here. BACK TO INDEX
5. Mixer Automation
Initially available only on profesional consoles, automation systems have evolved over the last 25 years,
becoming less expensive and more powerful in the process. Today there are moving fader and VCA based
systems available. Many only require insert points on the console and can be up and running fairly
quickly. Cheap VCA's and MIDI have brought automation to the masses.
a. SMPTE
We discussed SMPTE earlier in reference to locking up tape machines. It has also become the industry
standard for running automation systems. An obvious advantage is that one track of SMPTE is all you
have to give up on your multi-track to give you both synchronization and automation capabilities.
b. VCA vs. Moving Faders
I'm a big moving fader fan, particularly the "Flying Fader" system used on Neve VR consoles. Moving
fader systems utilize sophisticated mechanical, motorized faders running under the control of a computer
and software that keeps track of the physical position of the faders relative to the SMPTE timecode on the
tape. After recording a move, you simply rewind the tape and the faders replay your moves. With this type
of automation, the fader is actually in the audio path controlling the output of the signal.
Initially only available in very expensive hi-end consoles, motorized faders are showing up in low priced
consoles like the Yamaha 01v(about $1800), the O2R rev2(about $6000) and O3 consoles, Mackie has the
d8b (about $9000), the Panasonic DA7(about $5000), the TASCAM TM-d4000(about $4000) and the
Roland VM-72 system(about $4000). In 2000, Sony came out with its little "Oxford", the DMX-r100 for
about $25,000. None of these machines existed in 1995!
A VCA or "voltage controlled amplifier" is the heart of the SSL automation process. In this type of system
the fader controls the output of the VCA which is actually passing the signal. As in moving fader systems,
a computer and software keep track of the fader movements relative to SMPTE. Instead of the faders
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physically replaying your moves, their movement is represented on the computer monitor as a vertical
bar moving up and down. Many people don't like VCA's in the audio path, because they can color the
sound. This was more of a problem 25 years ago when VCA technology was young. The fact that SSL is
such a major player in big budget mixing says a lot about how little a problem that is. You can decide for
yourself.
c. MIDI Automation
The wide acceptance of MIDI and low cost, high quality VCA's has fueled the development of
inexpensive MIDI based automation systems. The mixing board gets a MIDI plug!
The TASCAM M-3700 is a good example of this approach. It's on-board system has a disc drive and small
LCD screen and allows you to automate channel volume, channel mute, monitor mute, EQ on/off and
effects send on/off. With additional software and an external computer you can have the moving bars on
the screen and some useful utility options.
Add-on outboard systems like the Mackie system utilize, a VCA package that ties in via the insert points
on your mixer and a smaller fader pack of 16 faders with mute buttons to record your moves on. This is all
translated into MIDI data which can be displayed on a Mac computer. Other systems come with software
or are already configured to work with several popular sequencer programs like Vision, Cubase and Logic.
These add-on VCA systems are a powerful and relatively inexpensive way to add sophisticated
automation capabilities to your mixer. BACK TO INDEX
6. Effects Devices
This is the cool stuff; reverbs, phase shifters, delays, chorus', harmonizers... and combinations never heard
before. These devices have come a long way in 20 years. Analog electronics and spring reverbs have
given way to very powerful digital multi-effects units with MIDI capabilities, memory for your favorite
patches and wide dynamic range. Five hundred dollars today, will buy some awesome sound power that
didn't even exist 20 years ago, at any price!
a. Reverb
Reverb units attempt to recreate the sound of a particular space. The way a "space" sounds is a product of
it's size, whether or not it's interior surfaces are hard and reflective or soft and absorbant, and how these
interior surfaces are arranged. All these factors interact to produce the reverberant sound. Two spaces can
have the same interior volume, but be shaped very differently and that makes all the difference.
The primary types of spaces are rooms, halls and plates but also can include chambers, churches, clubs
and any number of wild spaces. Some units give you a few parameters to tweak, others give you pages of
possibilities. All start with at least the "size" of the space. Other tweakable options include the volume and
intensity of "early reflections", the amount of pre-delay to the reverb, which delays the input send into the
reverb and "diffusion" and "depth" settings which have to do with how intensely the reverb spreads out in
the stereo field.
Special types of "reverse reverbs", where the sound envelope is turned around and ramps up in volume
rather than trailing off, are actually inspired by the analog trick of "backwards reverb". The important
distinction is that "reverse reverbs" occur after the sound just like regular reverbs. "Backwards reverb"
occurs before the sound and seems to ramp up to the sound.
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Backwards effects is an old analog recording technique. The record head is a fixed alignment of tracks, i.e.
1 - 8. When you turn the tape around on the tape machine, track 8 becomes 1, track 7 becomes 2, etc. If
you record your stereo backing vocals on tracks 1 and 2, for instance, then, at the end of the song, turn the
tape around and play it back, you will hear those vocals playing backwards on tracks 7 and 8. Send them
to a reverb while you're playing it backwards and record the reverb on 3 and 4. Think about it; the end of
the sung phrase hits the reverb first and it trails off after the start of the phrase.
When you turn the tape around and play it forward, the vocals will still be on 1 and 2, playing correctly,
but now there will be "backwards reverb" ramping up to the start of the sung phrases on track 3 and 4.
Check out Pink Floyd's "Dark Side of the Moon" for some tasty examples of backwards effects.
b. Echoes and Delays
An echo is an acoustic phenomenon where a sound is repeated. The classic example in mother nature is an
echo canyon. Delay units recreate this capability and give us control of several different parameters. The
two most basic are "delay time", the period between the input and the delayed signal's output, and
"feedback" or "regeneration" as it's sometimes called, which is how many times the signal repeats as it
fades away.
Generally, each successive repeat decreases in volume. When the feedback control is raised past a certain
point, the repeats get louder and louder. This is called "Runaway Feedback". Some units have a "Infinite
Hold" or "Freeze" feature which captures the input signal and keeps repeating it until you stop it. To the
untrained ear, discrete echoes start to be distinguished at about 20 milli-seconds. Delays lower than this
start to sound more like flangers and chorus effects. Most delay units also give you this capability.
The classic analog delay unit from the 1960's was the Maestro "Echoplex" and in the 1970's, the Roland
"Space Echo". Both used an endless tape loop and had a fixed record head and movable playback head.
You simply moved the playback head farther from the record head to achieve a longer delay. Early analog
electronic delays began to show up in the 1970's. The fidelity was somewhat limited and delay times only
went to 600 or 700 milli-seconds at most. Digital delays began to show up in the 1980's and delay times of
several seconds became more common. Modern units offer superb sonics, patch memories and MIDI
implementation.
c. Flanging, Chorus and Phasing
Like "reverse reverbs", these effects were inspired by analog recording techniques. The famous "slap-back
echo" of the 1950's and early 60's was created by sending the vocal to another tape machine in record and
mixing the signal coming off the repro-head back in with the original. The distance between the record
head and repro-head and the speed of the second machine determined the time of the "slap-back".
"Flanging" was a variation on this technique. The process starts similarly, but by rubbing the edge of the
flange of the tape reel on the 2nd machine ever so slightly, the characteristic "Flanging" sound was
produced. The time delays involved with flanging, chorus and phase shifting are usually well below 15
milli-seconds.
Digital delay units simulate these effects by incorporating an oscillator into their circuitry which allows
you to control the speed and depth of the signal being recombined with the original. Modern devices give
you mono and stereo flanging and chorus effects. Phase Shifters introduce slight time delays that also
change the phase of the signal being recombined. This gives the distinctive deep sweeping effect that they
are known for.
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d. Harmonizers and Exciters
Harmonizers are basically like digital delays except they allow us to tune the pitch of the delays. When
slightly detuned and added back in with the original signal, they are a useful tool for electronically
thickening vocals, guitars, etc. 'Smart' harmonizers are available now that can add designated pitches
according to different musical scales in their internal processors.
The most over-used effect of the 2000-2001 recording season has to be the Antares AutoTune. Originally
developed as a plugin for the ProTools recording format, its now available for most major software
recording packages and as a stand alone hardware unit, the ATR1. It can indeed repair pitch problems in a
performance, but it is now being used as an effect on pop, rap, dance and even country songs. When it is
overdriven, it resembles a vocorder but with more of the human voice intact.
The famous Aphex "aural exciter" was the first device of it's kind. The effect is often described as giving
"presence" to whatever it's used on. These kinds of devices all work slightly differently by adding and
subtracting harmonics, or readjusting the balance of phase relationships. It's definitely "voodoo", but very
useful at times. BACK TO INDEX
7. The MIDI Revolution
This truly has been a revolution of music power and access to it. It began in the early 1980's, as small
computer chips were finding their way into synthesizers. Several forward looking manufacturers of real
vision had the brilliant idea to establish a standard for communication between devices. This became the
first MIDI protocol. The MIDI standard has been expanding and improving since then and just about
every electronic musical device made today comes with MIDI plugs and some kind of implementation.
a. Sequencers
The first "sequencers" in the 1970's were analog. They were initially designed to playback a pattern that
you "stepped" in one note at a time and had a control to set the speed of the playback. They had a very
limited pattern memory and there was no standard way to lock any other device to them. MIDI sequencers
changed all that.
Hardware MIDI sequencers began to appear in the early 1980's. They were cheap and easy to use. The
Roland MC-500 and Yamaha QX series were very popular early models. As personal computers became
cheaper and more widely available, "software" sequencers began to appear. Dr.T for the Commodore C-64
was an early pioneer. Soon the Apple MacIntosh and Atari 1040 ST came on the scene and names like
"Performer", "Notator", "MasterTracks", "Vision" and "Cubase" became widely known software titles.
Software based sequencers are versatile, powerful devices which have several advantages over their
hardware cousins. Easy updates through software, a much larger monitor screen to view information on
and more intensive editing capabilities are just a few. The range of options and features is vast. A recent
innovation is the incorporating of digital recording into the sequencing program itself.
Confusion about Midi Clock and Midi Time Code (MTC) is very common. MTC and Midi Clock are
related but actually intended for different purposes. Midi clock came first and its principle role is to tell
listening midi devices what the tempo is(primarily sequencers and drum machines). Midi Song Pointer
came next and it tells other midi devices where bar 1 is, where bar 2 is, etc. As you can imagine, a high
degree of accuracy (we're talking milliseconds here) is needed for consistent control and lockup between
video decks, audio machines and midi equipment running together. Midi Time Code(MTC) was
developed to give midi devices an absolute reference point, much finer than bars or beats. Midi machine
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control(MMC) allows the sequencer to chase the audio recorder OR for the audio recorder to chase the
sequencer! Midi Machine Control uses MTC to keep things locked up to a very tight degree of resolution.
Most newer programs and MIDI devices support MTC and MMC. JL Cooper makes several different
boxes for communicating across formats. I use their DataMaster to read SMPTE off my 1/2" 16-track and
convert that to "ADAT speak" to lock my ADAT to the 16-track. The DataMaster also supports MTC and
MMC, but my sequencer doesn't. That's okay, I'll keep my Notator and Atari 1040-ST for now!
b. Synthesizers
Synthesizers have changed dramatically since computer chips began showing up in their circuitry in the
early 1980's. Before that, they were rather cumbersome machines given to tuning instability, usually two
voices at best and had to be reprogrammed manually for each new sound. The famous mini-moog and Arp
synthesizers were popular models in the 1970's.
The first truly polyphonic modern synthesizer with patch memory was the Sequential Circuits "Prophet 5"
which debuted in 1980. It had analog osciltors and their usual tuning problems but it's patch memory and
programming versatility revolutionized the industry and the use of synthesizers for live performance. For
the first time, keyboard players could change sounds with the touch of a button.
Digital oscilators soon followed and all variations of synthesis techniques were exploited in one form or
another. The "polyphony", or how many simultaneous voices could be produced by these units, also began
to increase. The MIDI protocol allowed you to have a different sound on each of 16 MIDI channels
limited only by the capabilities of your MIDI synthesizer. The Yamaha FB-01 was one of the earliest
synth "modules" to take advantage of this MIDI feature.
The Yamaha DX-7 became the most popular digital keyboard of the mid-1980's. Roland also produced
many popular keyboards in it's Jupiter and Juno series. Korg introduced onboard effects with it's DW-
6000 and DW-8000 synths. Korg hit another homerun with the M-1 synthesizer, introduced around 1990.
One of the first "workstation" designs, it combined sampling technology and synthesis to produce
breakthrough sonics along with an onboard sequencer and digital effects, to once again up the ante in the
synthesizer race. MIDI synthesizers keep getting more powerful all the time for less money, and that trend
continues.
c. Samplers
"Samplers" are like synthesizers in a lot of ways. In a synthesizer, oscilators produce the raw sound that is
then modified by filters and LFO's and sent through envelopes and amplifiers, etc. In a sampler, on the
other hand, the raw sound source can be anything that they sample. Then you can apply all the filters,
LFO's, envelopes and amplifiers to that.
The first keyboard samplers available in the early 1980's were the EMu Emulator series. With an
integrated 5 1/4" discdrive, they were big, heavy and expensive and awesome sounding. The Akai S-612
and Ensoniq Mirage were two of the first inexpensive rack mounted MIDI samplers. The prize for the first
inexpensive MIDI polyphonic keyboard sampler goes to the Casio FZ-1.
It appeared in 1985 and had 8 outputs and a standard 3 1/2" discdrive for saving. Akai made the very
popular S-900 rack mount samplers which evolved into the S-950. Then stereo samplers came along and
like everything else, they just keep getting more powerful and less expensive. Recent options available
include built-in CD Rom, SCSI hard disc and optical digital interfaces.
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d. Computers
Computers have become essentiel in modern music making. They are found in synthesizers, recording
devices, effects, automation and synchronization systems. They made the MIDI revolution possible.
The most popular music software in the 1980's was written for the Apple MacIntosh and Atari 1040-ST
computers. In Europe, the Atari was the dominant machine and C-Lab "Notator" and Steinberg "Cubase"
were two popular programs. In the US the Atari computers were much less expensive than the Macs but in
the early 90's, having business dificulties, they eventually disappeared from the market here. Atari still
enjoys a large presence in Europe and introduced the Falcon series there which includes 8-track digital
recording capabilities straight out of the box!
The Mac has always had a rabid following despite it's more expensive hardware cost. "Performer",
"Vision" and "MasterTracks Pro" were all strong sequencer packages written for it's unique operating
system. The hardware prices have moderated somewhat and it still enjoys a loyal user base and large share
of the music market.
The IBM was not that popular for music at first. Voyetra Systems had an early sequencing package for it
but it wasn't until "Windows" came along in the late 1980's that more software was written for it.
"Cakewalk" became a popular program for this platform in the early 1990's and soon others followed. E-
Magic's Logic Audio is a powerful digital recording and sequencing package that was originally written
for the Mac, but actually performs better on the Windows platform.
The real breakthrough for IBM has been it's rapidly expanding market share due to the popularity of
"Windows", the ever increasing power of the chips that drive the PC and the price differential between
PC's and Macs. The 386 gave way to the 486 which gave way to the Pentiums and their speed and
efficiency keeps growing. Apple had the edge on digital recording systems at one time but the
"Windows/IBM" platform has now become the dominant platform.
Hard disc recording systems are rapidly evolving and computers are either doing the recording and acting
as the "front-end" interface between the operator and recorder. The "SoundScape" hard disc recording
system from the UK uses a dedicated hardware recorder and a windows front-end. ProTools, which
previously only developed software for Macs, released the "Session 8", an 8-track hard disc recorder, for
the windows platform in 1993 and now ProTools is fully supported on the PC platform as well.
The computers have gotten powerful enough to handle it; so now you can sequence your synthesizers and
program your drums, then record your guitar amp and vocals into the computer and arrange the digitally
recorded tracks against the sequenced tracks all from within the same program. "Vision" for the Mac,
"Cakewalk Pro" and C-Lab "Logic" for the PC and "Cubase" for the Atari Falcon all have this capability.
Digital recording and editing are logical jobs for the computer and there are many systems taking
advantage of this power. Basically, the computer acts like a word processor for music. The material is
recorded into the computer and then you cut it up and rearrange it, EQ it, adjust it and put it back together
however you like. There are the usual complement of now standard AES/EBU digital ins and outs. Newer
designs are incorporating digital interfaces for the popular Alesis ADAT and Tascam DA-88 digital
recorders.
Another great use for a digital editing system is recording your mixes in pieces. If you don't have console
automation and the mix is too complicated, simply record your song in segments and edit it together.
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Several powerful digital editing programs are available now including "SAW Plus" and "Sound Forge".
BACK TO INDEX
8. Recording With Your PC
The explosion of recording hardware and software has finally collided with low prices and now everybody
wants to use their computer to make music. More than 1/2 the questions I get at 'Ask the Doctor' are now
computer recording related. The amount of sound power being offered from inside a PC today is
incredible.
The main problem is that it's too often a complicated process to get the hardware and software installed
properly and working correctly. However, once these "configuration" issues are solved, the PC recording
enviornment can be a lot of fun to use and can produce very high quality recordings.
to Mac users: I have no particular bone to pick with the Mac vs. the PC. I used Macs when I first started
working at Calliope Studios in NYC in 1986. Back then, for my own projects, the Atari was a lot cheaper
and did what I needed, so I used it. PC's and Mac's both crash from time to time and both have their
devotees. Use what you're comfortable with. The sound that comes out is only as good as the operator who
puts it in. "ProTools this and plugin-X" no more guarantees a hit song than using a "Strat through a
Marshall stack" means you'll sound just like Jimi Hendrix.
Remember, grasshopper, As you search for meaning in the musical universe, the computer is a tool that is
only as useful as he/she who uses it ...
THE SOFTWARE:
For starters, figure out what it is you want to do. Do you already have a demo setup, but need to edit and
compile your stereo mixes from DAT? or don't have a DAT and want to record your mixes directly into
the computer? or want to do multi-track recording into the computer, i.e. record a drummer, bass player
and guitarist all at the same time and mix on the computer? or you want to lock the computer to your
ADAT and use them both to record on? or you don't have any other equipment and want to do it all on the
computer?
Okay, you see that there are a lot of questions to answer before you buy the "box"! All of these software
companies have websites. Look at the software you might want to use and find out what the different
packages do. IMPORTANT!!! Every one of these software manufacturers lists the capabilities of each
piece of software as well as minimum PC capabilities required to run that software. I can guarantee you
that in the long run, you won't be happy with the way any music software runs on the "minimum machine"
listed in the software specs. Talk to a rep and make them tell you what it really needs to run on.
Recording Software comes in several flavors. There are 2-track stereo programs like Sonic Foundry's
Sound Forge. It's a very powerful 2-track editing package but mainly designed to work on one stereo song
at a time. With a Multi-track Recording program you could work on several songs at once, do cross fades
and overlap endings and beginnings, etc. or actually record and overdub parts as you would with any
multi-track recorder. There are lots of multi-track programs out there like Syntrillium's Cool Edit Pro, IQS
Saw Plus and SEK'd Samplitude, to name a few.
There are also powerful MIDI Sequencing programs with Digital Audio capabilities available like MOTU
Digital Performer, E-Magic Logic Audio and Opcode Vision DSP. With these you can build MIDI
sequences and record your audio and place the digital audio right onto the sequences. The audio becomes
another sound that is triggered along with the rest of the MIDI sequence.
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Here is where the capabilities of your soundcard come in. If you're only doing guitar and vocal
recording, then you'll probably be fine with a simple ANALOG L/R in/out soundcard. For overdubbing, it
must be "full duplex" which means the soundcard can input and output at the same time. A DIGITAL
connection on the soundcard means you can send your mix from inside the computer, out to a DAT
machine's DIGITAL input. To take advantage of the multi-track recording programs, you'll want multiple
ANALOG ins and outs and that's why they cost more. Look for the number of simultaneous record and
playback channels capabilities as well as the types of ANALOG and DIGITAL connections.
To burn cd's on your computer, you need cd-r software. A lot of software companies now have seperate
add-on cd-r software to go with their recording packages. In general, these are fairly sophisticated and
allow you to move id markers, change subcode info, etc. The cheapest stand alone cd-r software I've seen
is Adaptec's Easy CD-Pro. It came bundled with my cd-r. It simply allows you to arrange the order of your
audio files and burn the CD. Each seperate audio file gets an id on the CD. You can't place id's or access
subcode info.
Easy CD-Pro is also file backup software and this is really helpful when you've filled up that big hard
drive with audio files. Believe me, it will fill up faster than you can imagine! When I'm done with a
project on the computer, I save all the related files onto a cd-r and erase the old files on my hard drive.
When I need to work on it again, I reload the backed up files from cd-r and continue my slicing and
dicing.
So, you've checked out the software, hopefully even had a chance to speak with someone who is using it
or even better, actually seen it running. Now you need the hardware to run it on.
THE HARDWARE:
The Computer: Get the fastest Intel processor you can afford. The AMD and Cyrix chips are cheaper but
you might want to go with Intel. Specs I've seen from several companies won't guarantee software
performance on non-Intel chips, so read the fine print. Also check the motherboard for system bus speed
(66mhz on older models, 100mhz and 133 mhz on newer ones), amount and type of onboard RAM it can
hold(more is better) and number of open slots for PCI/ISA cards(more is better). If you have older ISA
cards you want to use, double check because many new motherboards only offer PCI.
The RAM: Stuff at least 128 meg of RAM in there, 256 meg or more would be even better. It used to be
30 and 72-pin 60ns memory chips. Newer motherboards can take 168-pin SDRAM memory chips that run
at 7ns on the 100mhz bus machines. Check the motherboard manuals for the type, speed and physical
parameters of the memory you should use. Some machines take (2) pairs, some can handle different sizes
(a 64 and 128), some can't.
The Monitor: All of these software recording packages suffer from the same problem. They fill the screen
with too much information constantly. 17" monitors are getting down below $300, so go for a 19" or 21" if
you can. You'll thank me later. As of 2001, 15" flat screen monitors offer almost the same screen real
estate as a 17" monitor, and are in the $600 range. They take up a lot less space and give off lots less heat.
PCI/ISA slots: Most music software is now taking advantage of the PCI slots. You'll want at least (4) of
them. Some PCI cards are long and some are short, so the interior physical dimensions of your computer
enclosure could come into play. Get a tower enclosure if possible, it'll give you room to add hard drives
and more stuff later on.
Power Supply: You'll want the biggest you can get because you'll have lots of peripherals stuffed in there.
Definitely opt for the quietest one you can get as well. Those fan(s) can be a problem.
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Hard Drives: Get a seperate Hard Drive(s) that will only be used for digital audio. Audio files are huge
and you don't want to be recording on the same hard drive that your operating system is on. IDE drives
will work and the new Ultra-ATA (UDMA)IDE drives are fast(a 66MB/sec and even newer 100MB/sec
version) and very cheap.
A PC motherboard has sockets for 4 IDE drives (Primary master and slave AND Secondary master and
slave) although some may only allow 2 of the faster UDMA drive types. To connect additional hard drives
takes a little work. You'll have to make sure your system BIOS can recognize the drive. There's usually
some kind of message when you boot up about entering "setup", press F1 or something like that. The
instructions with the drive and your motherboard will help you with these settings. You'll also need to set
some pins on the drive itself to set it as the master or slave.
I'm using 3 hard drives on my current system. One is dedicated to the operating system and programs. The
other 2 are for digital audio. One is an EIDE 1.6 gig and the other is a UDMA 6.4 gig. The EIDE drive
works fine, its just slower. CHECK THE MOTHERBOARD SPECS! A lot of software problems are
actually due to hardware issues.
It used to be that for more than 4 connection capabilities, your only choice was a SCSI card which allows
at least (7) SCSI devices to be connected at once. SCSI comes in several flavors and is a little faster, but
SCSI peripherals generally cost 2 or 3 times as much as their IDE cousins. The chart below shows the
respective transfer speeds.
IDE/UltraDMA SCSI
IDE/ATA 2.1 - 8.3 MB/sec SCSI 5 MB/sec
EIDE 11.1 - 16.6 MB/sec Fast SCSI 10 MB/sec
Ultra ATA(UDMA) 33.3 MB/sec Fast Wide SCSI 20 MB/sec
New Ultra-DMA 66 MB/sec Ultra SCSI 20 MB/sec
New Ultra-DMA II 100 MB/sec Ultra 2 SCSI 40 MB/sec
USB is now an option to SCSI. Theoretically, you can daisy chain 127 devices with USB. There's no
special card needed because it's a built-in format on most newer computers (post 1999) and its plenty fast
to handle digital audio. It's convenient because the USB peripherals are "hot-swappable" (meaning you
can plug and unplug them with the computer on) and the cabling requirements are simpler than with SCSI
devices. USB hard drives, cd-r's, cd-rw's and removeable drives are all available.
Tascam and Event both have come out with small digital audio mixers that use the USB interface. They're
both about $600 and provide 8 faders, EQ, digital input and output, etc. When you get tired of recording
with the mouse, you can check out one of these as an interface between you and your favorite recording
software.
The newest audio interface on the block is FireWire and this one will fuel the next big audio hardware
revolution. No PCI cards, just plug in and start recording. Its right on the motherboard like USB and ready
to rock and roll. Devices that take advantage of FireWire are just starting to show up in 2001.
Removable Drives: A standard Zip drive cartridge will hold about 10 minutes of 44.1 stereo digital audio.
A 1 meg Jazz or Syquest Sparq drive cartridge holds about 100 minutes of 44.1 stereo digital audio. If
you're doing a lot of different projects, a removeable storage medium can be very helpful. The zip
cartridges are about $10 apiece, Jazz carts are $90-100 each and Sparq carts are $33 apiece.
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Unfortunately, Syquest is out of business as of summer '99 , so Iomega doesn't really have any
competition and can charge what they want. There is a new removeable drive out there called the Orbit
that looks a lot like the Syquest Sparq. It holds 2.2gig per cartridge and they're also about $35 apiece.
CD-r's and CD-rw's: 12x speed cd-r's are now in the $300 range! They can be SCSI or IDE as well. Bulk
cd-r's on a spindle are down to $.20 each. Get a decent inkjet, do your own labels and make your own cd's
one at a time; voila, instant record label. I picked up an 8x SCSI cd-r for $200 in Nov.99. I actually have a
2x SCSI cd-r and the 8x SCSI cd-r both installed on my system. I have some old backup software that
works with the 2x cd-r and I use the 8x for burning cd's. I love burning a 50 minute cd in 8 minutes!
CD-rw is great for backing up your files. It looks like another hard drive to the computer and you can
reuse the medium. The physical cd's are more expensive, but it's a no-brainer.
The Soundcard: This is the device that gets the sound into and out of the computer. There are lots of
soundcards out there and you need to pick the right one for doing what you want to do. Again, go to the
maufacturers' websites. Make sure the software that you have picked will work with the soundcard you're
looking at. The software should list soundcards that have been tested and work correctly with it.
Don't get a card that only has digital in/outs unless you have a DAT machine or other box that reads a
digital signal. There are two flavors of digital in/out. One is S/PDIF which uses an RCA style connector
(like your cassette deck) and the other is AES/EBU which uses a cannon connector (like a microphone
cable). Some cards have one, some have both. You can't plug them into your home stereo and hear them!
Whether its simply for stereo editing or full blown multi-track recording, get a soundcard with at least a
stereo analog monitor output so when all the digital bells and whistles quit on you, you can plug this into
your stereo and hear what the hell is going on OR not going on!
A Word About Wordclock: If you're only using the computer to record and mix on, this doesn't apply to
you. If on the other hand, you're planning on using the computer with other digital divices such as ADAT's
or DA-88's, or a digital mixer like the Panasonic DA7 or Yamaha 02R, then this is very important to you.
All digital devices have computers inside them running their own internal digital clocks. To work properly
together, there has to be a MASTER WORDCLOCK device with all the others being connected to it.
When you start to interconnect them and move the digital audio between devices, and don't properly
connect the wordclock ins and outs, these different digital clocks begin to cause problems which can
include pops and clicks, random noise, timing discrepancies, audio drift or no audio at all. Furthermore,
different combinations of digital equipment will require some experimentation as to which particular
device should be the wordclock master.
The Drivers: These are little software programs that communicate with the operating system, be it Win95,
Win98, NT or Win3.11 and facilitate the smooth interaction between your H/W and S/W. Win95 had a
ver. 1 and a version 2 as did Windows 98 and now we have Windows 2000 to contend with. You need to
make sure that your software and hardware BOTH are supported by whatever version of Windows is on
your computer.
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NT is particularly unforgiving in this area because it does not support plug and play. NT is very robust
but the hardware has to be approved specifically for the NT platform. Updated drivers are always
promised but if they aren't available, you're out of luck.
COMPATIBILITY ISSUES: Most software manufacturers will list systems and hardware that they have
tested to work with their products. Find this information and use it. It could save you a lot of headaches.
Particularly because the PC is a "roll your own" box, the money you save building a system yourself, is
worthless if the software won't perform properly on it.
So let's review. To get into the computer recording game, you'll need:
COMPUTER HARDWARE
the computer $1000-2000
the monitor $ 200-800
the soundcard $ 350-1000
dedicated hard drive(s) $ 200-600
cd-r/cd-rw $ 200-600
recording software $ 100-800
MIDI sequencing software (?) $ 100-800
cd-r/cd-rw software (?) $ 100-500
SCSI card (?) $ 120-300
UltraDMA-66/100 card (?) $ 100
Removeable drive
AUDIO HARDWARE
(?) $ 300
microphones/preamps (?)
external signal processors (?)
external mixer (?)
speakers (?)
external recorders (?)
external digital
synchronization
(?)
This is a substantial investment and as you can see, more involved than just buying a computer. If you're
not a gear slut/techie, then save yourself a lot of aggravation and have the computer and all it's hard drives,
removeable drives, cd-roms and cd-r's put together and configured by the same outfit. Remember to do
your homework and read the fine print before you buy anything. Good luck and if you have any questions,
just ask the doctor! BACK TO INDEX
Microphones
• I. How They Work.
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• II. Specifications.
• III. Pick Up Patterns
• IV. Typical Placement
• V. The Microphone Mystique
I. How They Work.
A microphone is an example of a transducer, a device that changes information from one form to another.
Sound information exists as patterns of air pressure; the microphone changes this information into patterns
of electric current. The recording engineer is interested in the accuracy of this transformation, a concept he
thinks of as fidelity.
A variety of mechanical techniques can be used in building microphones. The two most commonly
encountered in recording studios are the magneto-dynamic and the variable condenser designs.
THE DYNAMIC MICROPHONE.
In the magneto-dynamic, commonly called dynamic, microphone, sound waves cause movement of a thin
metallic diaphragm and an attached coil of wire. A magnet produces a magnetic field which surrounds the
coil, and motion of the coil within this field causes current to flow. The principles are the same as those
that produce electricity at the utility company, realized in a pocket-sized scale. It is important to remember
that current is produced by the motion of the diaphragm, and that the amount of current is determined by
the speed of that motion. This kind of microphone is known as velocity sensitive.
THE CONDENSER MICROPHONE.
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In a condenser microphone, the diaphragm is mounted close to, but not touching, a rigid backplate. (The
plate may or may not have holes in it.) A battery is connected to both pieces of metal, which produces an
electrical potential, or charge, between them. The amount of charge is determined by the voltage of the
battery, the area of the diaphragm and backplate, and the distance between the two. This distance changes
as the diaphragm moves in response to sound. When the distance changes, current flows in the wire as the
battery maintains the correct charge. The amount of current is essentially proportioinal to the displacement
of the diaphragm, and is so small that it must be electrically amplified before it leaves the microphone.
A common varient of this design uses a material with a permanently imprinted charge for the diaphragm.
Such a material is called an electret and is usually a kind of plastic. (You often get a piece of plastic with a
permanent charge on it when you unwrap a record. Most plastics conduct electricity when they are hot but
are insulators when they cool.) Plastic is a pretty good material for making diaphragms since it can be
dependably produced to fairly exact specifications. (Some popular dynamic microphones use plastic
diaphragms.) The major disadvantage of electrets is that they lose their charge after a few years and cease
to work.
II. Specifications
There is no inherent advantage in fidelity of one type of microphone over another. Condenser types
require batteries or power from the mixing console to operate, which is occasionally a hassle, and
dynamics require shielding from stray magnetic fields, which makes them a bit heavy sometmes, but very
fine microphones are available of both styles. The most important factor in choosing a microphone is how
it sounds in the required application. The following issues must be considered:
Sensitivity.
This is a measure of how much electrical output is produced by a given sound. This is a vital specification
if you are trying to record very tiny sounds, such as a turtle snapping its jaw, but should be considered in
any situation. If you put an insensitive mic on a quiet instrument, such as an acoustic guitar, you will have
to increase the gain of the mixing console, adding noise to the mix. On the other hand, a very sensitive mic
on vocals might overload the input electronics of the mixer or tape deck, producing distortion.
Overload characteristics.
Any microphone will produce distortion when it is overdriven by loud sounds. This is caused by varous
factors. With a dymanic, the coil may be pulled out of the magnetic field; in a condenser, the internal
amplifier might clip. Sustained overdriving or extremely loud sounds can permanently distort the
diaphragm, degrading performance at ordinary sound levels. Loud sounds are encountered more often than
you might think, especially if you place the mic very close to instruments. (Would you put your ear in the
bell of a trumpet?) You usually get a choice between high sensitivity and high overload points, although
occasionally there is a switch on the microphone for different situations.
Linearity, or Distortion.
This is the feature that runs up the price of microphones. The distortion characteristics of a mic are
determined mostly by the care with which the diaphragm is made and mounted. High volume production
methods can turn out an adequate microphone, but the distortion performance will be a matter of luck.
Many manufacturers have several model numbers for what is essentially the same device. They build a
batch, and then test the mics and charge a premium price for the good ones. The really big names throw
away mic capsules that don't meet their standards. (If you buy one Neumann mic, you are paying for five!)
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No mic is perfectly linear; the best you can do is find one with distortion that complements the sound you
are trying to record. This is one of the factors of the microphone mystique discussed later.
Frequency response.
A flat frequency response has been the main goal of microphone companies for the last three or four
decades. In the fifties, mics were so bad that console manufacturers began adding equalizers to each input
to compensate. This effort has now paid off to the point were most professional microphones are
respectably flat, at least for sounds originating in front. The major exceptions are mics with deliberate
emphasis at certain frequencies that are useful for some applications. This is another part of the
microphone mystique. Problems in frequency response are mostly encountered with sounds originating
behind the mic, as discussed in the next section.
Noise.
Microphones produce a very small amount of current, which makes sense when you consider just how
light the moving parts must be to accurately follow sound waves. To be useful for recording or other
electronic processes, the signal must be amplified by a factor of over a thousand. Any electrical noise
produced by the microphone will also be amplified, so even slight amounts are intolerable. Dynamic
microphones are essentially noise free, but the electronic circuit built into condensor types is a potential
source of trouble, and must be carefully designed and constructed of premium parts.
Noise also includes unwanted pickup of mechanical vibration through the body of the microphone. Very
sensitive designs require elastic shock mountings, and mics intended to be held in the hand need to have
such mountings built inside the shell.
The most common source of noise associated with microphones is the wire connecting the mic to the
console or tape deck. A mic preamp is very similar to a radio reciever, so the cable must be prevented
from becoming an antenna. The basic technique is to surround the wires that carry the current to and from
the mic with a flexible metallic shield, which deflects most radio energy. A second technique, which is
more effective for the low frequency hum induced by the power company into our environment, is to
balance the line:
Current produced by the microphone will flow down one wire of the twisted pair, and back along the other
one. Any current induced in the cable from an outside source would tend to flow the same way in both
wires, and such currents cancel each other in the transformers. This system is expensive.
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Microphone Levels
As I said, microphone outputs are of necessity very weak signals, generally around -60dBm. (The
specification is the power produced by a sound pressure of 10 uBar) The output impedance will depend on
whether the mic has a transformer balanced output . If it does not, the microphone will be labeled "high
impedance" or "hi Z" and must be connected to an appropriate input. The cable used must be kept short,
less than 10 feet or so, to avoid noise problems.
If a microphone has a transformer, it will be labeled low impedance, and will work best with a balanced
input mic preamp. The cable can be several hundred feet long with no problem. Balanced output, low
impedance microphones are expensive, and generally found in professonal applications. Balanced outputs
must have three pin connectors ("Cannon plugs"), but not all mics with those plugs are really balanced.
Microphones with standard or miniature phone plugs are high impedance. A balanced mic can be used
with a high impedance input with a suitable adapter.
You can see from the balanced connection diagram that there is a transformer at the input of the console
preamp. (Or, in lieu of a transformer, a complex circuit to do the same thing.) This is the most significant
difference between professional preamplifiers and the type usually found on home tape decks. You can
buy transformers that are designed to add this feature to a consumer deck for about $20 each. (Make sure
you are getting a transformer and not just an adapter for the connectors.) With these accessories you can
use professional quality microphones, run cables over a hundred feet with no hum, and because the
transformers boost the signal somewhat, make recordings with less noise. This will not work with a few
inexpensive cassette recorders, because the strong signal causes distortion. Such a deck will have other
problems, so there is little point trying to make a high fidelity recording with it anyway.
III. Pick Up Patterns
Many people have the misconception that microphones only pick up sound from sources they are pointed
at, much as a camera only photographs what is in front of the lens. This would be a nice feature if we
could get it, but the truth is we can only approximate that action, and at the expense of other desirable
qualities.
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MICROPHONE PATTERNS
These are polar graphs of the output produced vs. the angle of the sound source. The output is represented
by the radius of the curve at the incident angle.
Omni
The simplest mic design will pick up all sound, regardless of its point of origin, and is thus known as an
omnidirectional microphone. They are very easy to use and generally have good to outstanding frequency
response. To see how these patterns are produced, here's a sidebar on directioal microphones.
Bi-directional
It is not very difficult to produce a pickup pattern that accepts sound striking the front or rear of the
diaphragm, but does not respond to sound from the sides. This is the way any diaphragm will behave if
sound can strike the front and back equally. The rejection of undesired sound is the best achievable with
any design, but the fact that the mic accepts sound from both ends makes it difficult to use in many
situations. Most often it is placed above an instrument. Frequency response is just as good as an omni, at
least for sounds that are not too close to the microphone.
Cardioid
This pattern is popular for sound reinforcement or recording concerts where audience noise is a possible
problem. The concept is great, a mic that picks up sounds it is pointed at. The reality is different. The first
problem is that sounds from the back are not completely rejected, but merely reduced about 10-30 dB.
This can surprise careless users. The second problem, and a severe one, is that the actual shape of the
pickup pattern varies with frequency. For low frequencies, this is an omnidirectional microphone. A mic
that is directional in the range of bass instruments will be fairly large and expensive. Furthermore, the
frequency response for signals arriving from the back and sides will be uneven; this adds an undesired
coloration to instruments at the edge of a large ensemble, or to the reverberation of the concert hall.
A third effect, which may be a problem or may be a desired feature, is that the microphone will emphasize
the low frequency components of any source that is very close to the diaphragm. This is known as the
"proximity effect", and many singers and radio announcers rely on it to add "chest" to a basically light
voice. Close, in this context, is related to the size of the microphone, so the nice large mics with even back
and side frequency response exhibit the strongest presence effect. Most cardioid mics have a built in
lowcut filter switch to compensate for proximity. Missetting that switch can cause hilarious results.
Bidirectional mics also exhibit this phenomenon.
Tighter Patterns
It is posible to exaggerate the directionality of cardioid type microphones, if you don't mind exaggerating
some of the problems. The Hypercardioid pattern is very popular, as it gives a better overall rejection and
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flatter frequency response at the cost of a small back pickup lobe. This is often seen as a good
compromise between the cardioid and bidirectional patterns. A "shotgun" mic carries these techniques to
extremes by mounting the diaphragm in the middle of a pipe. The shotgun is extremely sensitive along the
main axis, but posseses pronounced extra lobes which vary drastically with frequency. In fact, the
frequency response of this mic is so bad it is usually electronically restricted to the voice range, where it is
used to record dialogue for film and video.
Stereo microphones
You don't need a special microphone to record in stereo, you just need two (see below). A so called stereo
microphone is really two microphones in the same case. There are two kinds: extremely expensive
professional models with precision matched capsules, adjustable capsule angles, and remote switching of
pickup patterns; and very cheap units (often with the capsules oriented at 180 deg.) that can be sold for
high prices because they have the word stereo written on them.
IV. Typical Placement
Single microphone use
Use of a single microphone is pretty straightforward. Having chosen one with appropriate sensitivity and
pattern, (and the best distortion, frequency response, and noise characteristics you can afford), you simply
mount it where the sounds are. The practical range of distance between the instrument and the microphone
is determined by the point where the sound overloads the microphone or console at the near end, and the
point where ambient noise becomes objectionable at the far end. Between those extremes it is largely a
matter of taste and experimentation.
If you place the microphone close to the instrument, and listen to the results, you will find the location of
the mic affects the way the instrument sounds on the recording. The timbre may be odd, or some notes
may be louder than others. That is because the various components of an instrument's sound often come
from different parts of the instrument body (the highest note of a piano is nearly five feet from the lowest),
and we are used to hearing an evenly blended tone. A close in microphone will respond to some locations
on the instrument more than others because the difference in distance from each to the mic is
proportionally large. A good rule of thumb is that the blend zone starts at a distance of about twice the
length of the instrument. If you are recording several instruments, the distance between the players must
be treated the same way.
If you place the microphone far away from the instrument, it will sound as if it is far away from the
instrument. We judge sonic distance by the ratio of the strength of the direct sound from the instrument
(which is always heard first) to the strength of the reverberation from the walls of the room. When we are
physically present at a concert, we use many cues beside the sounds to keep our attention focused on the
performance, and we are able to ignore any distractions there may be. When we listen to a recording, we
don't have those visual clues to what is happening, and find anything extraneous that is very audible
annoying. For this reason, the best seat in the house is not a good place to record a concert. On the other
hand, we do need some reverberation to appreciate certain features of the music. (That is why some types
of music sound best in a stone church) Close microphone placement prevents this. Some engineers prefer
to use close miking techniques to keep noise down and add artificial reverberation to the recording, others
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solve the problem by mounting the mic very high, away from audience noise but where adequate
reverberation can be found.
Stereo
Stereo sound is an illusion of spaciousness produced by playing a recording back through two speakers.
The success of this illusion is referred to as the image. A good image is one in which each instrument is a
natural size, has a distinct location within the sound space, and does not move around. The main factors
that establish the image are the relative strength of an instrument's sound in each speaker, and the timing
of arrival of the sounds at the listener's ear. In a studio recording, the stereo image is produced artificially.
Each instrument has its own microphone, and the various signals are balanced in the console as the
producer desires. In a concert recording, where the point is to document reality, and where individual
microphones would be awkward at best, it is most common to use two mics, one for each speaker.
Spaced microphones
The simplest approach is to assume that the speakers will be eight to ten feet apart, and place two
microphones eight to ten feet apart to match. Either omnis or cardioids will work. When played back, the
results will be satisfactory with most speaker arrangements. (I often laugh when I attend concerts and
watch people using this setup fuss endlessly with the precise placement of the mics. This technique is so
forgiving that none of their efforts will make any practical difference.)
The big disavantage of this technique is that the mics must be rather far back from the ensemble- at least
as far as the distance from the leftmost performer to the rightmost. Otherwise, those instruments closest to
the microphones will be too prominent. There is usually not enough room between stage and audience to
achieve this with a large ensemble, unless you can suspend the mics or have two very tall stands.
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Coincident cardioids
There is another disadvantage to the spaced technique that appears if the two channels are ever mixed
together into a monophonic signal. (Or broadcast over the radio, for similar reasons.) Because there is a
large distance between the mics, it is quite possible that sound from a particular instrument would reach
each mic at slightly different times. (Sound takes 1 millisecond to travel a foot.) This effect creates phase
differences between the two channels, which results in severe frequency response problems when the
signals are combined. You seldom actually lose notes from this interference, but the result is an uneven,
almost shimmery sound. The various coincident techniques avoid this problem by mounting both mics in
almost the same spot.
This is most often done with two cardioid microphones, one pointing slightly left, one slightly right. The
microphones are often pointing toward each other, as this places the diaphragms within a couple of inches
of each other, totally eliminating phase problems. No matter how they are mounted, the microphone that
points to the left provides the left channel. The stereo effect comes from the fact that the instruments on
the right side are on-axis for the right channel microphone and somewhat off-axis (and therefore reduced
in level) for the other one. The angle between the microphones is critical, depending on the actual pickup
pattern of the microphone. If the mics are too parallel, there will be little stereo effect. If the angle is too
wide, instruments in the middle of the stage will sound weak, producing a hole in the middle of the image.
[Incidentally, to use this technique, you must know which way the capsule actually points. There are some
very fine German cardioid microphones in which the diaphragm is mounted so that the pickup is from the
side, even though the case is shaped just like many popular end addressed models. (The front of the mic in
question is marked by the trademark medallion.) I have heard the results where an engineer mounted a pair
of these as if the axis were at the end. You could hear one cello player and the tympani, but not much
else.]
You may place the microphones fairly close to the instruments when you use this technique. The problem
of balance between near and far instruments is solved by aiming the mics toward the back row of the
ensemble; the front instruments are therefore off axis and record at a lower level. You will notice that the
height of the microphones becomes a critical adjustment.
M.S.
The most elegant approach to coincident miking is the M.S. or middle-side technique. This is usually done
with a stereo microphone in which one element is omnidirectional, and the other bidirectional. The
bidirectional element is oriented with the axis running parallel to the stage, rejecting sound from the
center. The omni element, of course, picks up everything. To understand the next part, consider what
happens as instrument is moved on the stage. If the instrument is on the left half of the stage, a sound
would first move the diaphragm of the bidirectional mic to the right, causing a positive voltage at the
output. If the instrument is moved to center stage, the microphone will not produce any signal at all. If the
instrument is moved to the right side, the sound would first move the diaphragm to the left, producing a
negative volage. You can then say that instruments on one side of the stage are 180 degrees out of phase
with those on the other side, and the closer they are to the center, the weaker the signal produced.
Now the signals from the two microphones are not merely kept in two channels and played back over
individual speakers. The signals are combined in a circuit that has two outputs; for the left channel output,
the bidirectional output is added to the omni signal. For the right channel output, the bidirectional output is
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subtracted from the omni signal. This gives stereo, because an instrument on the right produces a
negative signal in the bidirectional mic, which when added to the omni signal, tends to remove that
instrument, but when subtracted, increases the strength of the instrument. An instrument on the left suffers
the opposite fate, but instruments in the center are not affected, because their sound does not turn up in the
bidirectional signal at all.
M.S. produces a very smooth and accurate image, and is entirely mono compatabile. The only reason it is
not used more extensively is the cost of the special microphone and decoding circuit, well over $1,000.
Large ensembles
The above techniques work well for concert recordings in good halls with small ensembles. When
recording large groups in difficult places, you will often see a combination of spaced and coincident pairs.
This does produce a kind of chorusing when the signals are mixed, but it is an attractive effect and not
very different from the sound of string or choral ensembles any way. When balance between large sections
and soloists cannot be acheived with the basic setup, extra microphones are added to highlight the weaker
instruments. A very common problem with large halls is that the reverberation from the back seems late
when compared to the direct sound taken at the edge of the stage. This can be helped by placing a mic at
the rear of the audience area to get the ambient sound into the recording sooner.
Studio techniques
A complete description of all of the procedures and tricks encountered in the recording studio would fill
several books. These are just a few things you might see if you dropped in on the middle of a session.
Individual mics on each instrument.
This provides the engineer with the ability to adjust the balance of the instruments at the console, or, with
a multitrack recorder, after the musicians have gone home. There may be eight or nine mics on the drum
set alone.
Close mic placement.
The microphones will usually be placed rather close to the instruments. This is partially to avoid problems
that occur when an instrument is picked up in two non-coincident mics, and partially to modify the sound
of the instruments (to get a "honky-tonk" effect from a grand piano, for instance).
Acoustic fences around instruments, or instruments in separate rooms.
The interference that occurs when when an instrument is picked up by two mics that are mixed is a very
serious problem. You will often see extreme measures, such as a bass drum stuffed with blankets to muffle
the sound, and then electronically processed to make it sound like a drum again.
Everyone wearing headphones.
Studio musicians often play to "click tracks", which are not recorded metronomes, but someone tapping
the beat with sticks and occasionally counting through tempo changes. This is done when the music must
be synchronized to a film or video, but is often required when the performer cannot hear the other
musicians because of the isolation measures described above.
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20 or 30 takes on one song.
Recordings require a level of perfection in intonation and rhythm that is much higher than that acceptable
in concert. The finished product is usually a composite of several takes.
Pop filters in front of mics.
Some microphones are very sensitive to minor gusts of wind--so sensitive in fact that they will produce a
loud pop if you breath on them. To protect these mics (some of which can actually be damaged by
blowing in them) engineers will often mount a nylon screen between the mic and the artist. This is not the
most common reason for using pop filters though:
Vocalists like to move around when they sing; in particular, they will lean into microphones. If the singer
is very close to the mic, any motion will produce drastic changes in level and sound quality. (You have
seen this with inexpert entertainers using hand held mics.) Many engineers use pop filters to keep the artist
at the proper distance. The performer may move slightly in relation to the screen, but that is a small
proportion of the distance to the microphone.
V. The Microphone Mystique
There is an aura of mystery about microphones. To the general public, a recording engineer is something
of a magician, privy to a secret arcana, and capable of supernatural feats. A few modern day engineers
encourage this attitude, but it is mostly a holdover from the days when studio microphones were expensive
and fragile, and most people never dealt with any electronics more complex than a table radio. There are
no secrets to recording; the art is mostly a commonsense application of the principles already discussed in
this paper. If there is an arcana, it is an accumulation of trivia achieved through experience with the
following problems:
Matching the microphone to the instrument.
There is no wrong microphone for any instrument. Every engineer has preferences, usually based on mics
with which he is familiar. Each mic has a unique sound, but the differences between good examples of any
one type are pretty minor. The artist has a conception of the sound of his instrument, (which may not be
accurate) and wants to hear that sound through the speakers. Frequency response and placement of the
microphone will affect that sound; sometimes you need to exaggerate the features of the sound the client is
looking for.
Listening the proper way.
It is easy to forget that the recording engineer is an illusionist- the result will never be confused with
reality by the listener. Listeners are in fact very forgiving about some things. It is important that the
engineer be able to focus his attention on the main issues and not waste time with interesting but minor
technicalities. It is important that the engineer know what the main issues are. An example is the
noise/distortion tradeoff. Most listeners are willing to ignore a small amount of distortion on loud passages
(in fact, they expect it), but would be annoyed by the extra noise that would result if the engineer turned
the recording level down to avoid it. One technique for encouraging this attention is to listen to recordings
over a varitey of sound systems, good and bad.
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Learning for yourself.
Many students come to me asking for a book or a course of study that will easily make them a member of
this elite company. There are books, and some schools have courses in recording, but they do not supply
the essential quality the professional recording engineer needs, which is experience.
A good engineer will have made hundreds of recordings using dozens of different microphones. Each
session is an opportunity to make a new discovery. The engineer will make careful notes of the setup, and
will listen to the results many times to build an association between the technique used and the sound
achieved. Most of us do not have access to lots of professional microphones, but we could probably afford
a pair of general purpose cardioids. With about $400 worth of mics and a reliable tape deck, it is possible
to learn to make excellent recordings. The trick is to record everything that will sit still and make noise,
and study the results: learn to hear when the mic is placed badly and what to do about it. When you know
all you can about your mics, buy a different pair and learn those. Occasionally, you will get the
opportunity to borrow mics. If possible, set them up right alongside yours and make two recordings at
once. It will not be long before you will know how to make consistently excellent recordings under most
conditions.
Peter Elsea 1996
MESSING WITH THE SOUNDS
The devices collectively known as audio processors were originally added to the recording studio to allow
compensation for frequency response or dynamic range problems in the equipment. When carefully used,
they can add to the fidelity of the sound, and occasionally improve on reality. If improperly used, they can
seriously degrade the sound or even produce laughable results. It is possible to use audio processors to
manipulate sounds into unrecognizablilty, so naturally they are popular with composers of electronic
music.
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EQUALIZATION (E.Q.)
An equalizer is a device that can alter the spectral content of a signal. This can be done with any circuit
that has an adjustable frequency response, the most familiar being the tone controls on a home stereo set.
These tone controls typically affect the amplitude in two frequency regions, the treble and bass. This is
sufficient for the minor changes the end user may wish to make in the program, but the recording engineer
needs more flexibility and coverage of the entire audio spectrum. The complete studio will have complex
e.q. systems that might include 30 or more regions of control. Many of these machines have sliders to
adjust the amplitude of each band, and those sliders are laid out in such a way that their positions visually
indicate the frequency response. This feature gives rise to the name GRAPHIC EQUALIZER.
A 30 band equalizer is going to be an expensive device, simply because of the sheer duplication of
circuitry. Much of this circuitry is wasted most of the time because typical use only involves adjustment to
two or three bands while the others are left alone. An equalizer with bands of adjustable frequency is a
more economical approach to the problem, because only three to five circuits are necessary. Such an
equalizer is called a PARAMETRIC EQUALIZER. Parametric equalizers typically have three controls for
each band, allowing adjustment of frequency, amplitude, and width of the band.
Fig. 1. Equalizers
FILTERS
Equalizers are deliberately designed to create fairly minor changes in the signal. For more drastic effects,
such as removing some region of the signal entirely, a FILTER is required. A filter is a circuit that sharply
reduces the amplitude of signals of frequency outside of specified limits. The unaffected region is called
the PASSBAND, and the filter type is named after the passband as low-pass, high-pass, or band-pass. The
point where the signal attenuation just becomes noticeable (a reduction of 3 dB) is termed the CUTOFF
FREQUENCY. A low-pass filter with a cutoff of 500 hz will attenuate any signal of frequency above
500hz.
Fig. 2 Filter response curves
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The attenuation provided by a filter is never absolute. A graph of the frequency response of the filter
shows that the amplitude of a signal decreases as the frequency moves beyond the cutoff. The actual rate
of this decrease is a parameter of filter design called the SLOPE. For arcane reasons the slope is always
some multiple of 6 dB of attenuation per octave of signal frequency change. (A 6dB per octave filter is
really not much more than an equalizer, and the drastic effects used in electronic music generally require
24 dB per octave circuits.) The slope is a general description of response at some remove from the cutoff.
Various circuits differ in the shape of the curve near cutoff, and the possible shapes have names such as
Bessel and Chebychev, after the originators of the math formulas involved.
Another design parameter which affects the shape of the filter curve is known as "Q". The derivation of Q
is too complex for this discussion, but it is useful to know that filters with a high value of Q have an
amplitude bump near the cutoff frequency and have a tendency to oscillate at that frequency.
Fig. 3 Bandpass response curve
Bandpass filters have two cutoff frequencies. The difference between these frequencies is the
BANDWIDTH, and the mean of the two is the CENTER FREQUENCY. Bandpass filters are sometimes
encountered in large groups of fixed frequency circuits similar to graphic equalizers. These filter banks are
often called 1/2 octave or 1/3 octave filters after the spacing of the filter bands. Such devices used to be
the mainstay of many tape music studios before stable synthesizer filters and parametric equalizers were
available. The characteristic pitches associated with those machines is almost a trademark of early 60's
tape music.
The 1/3 octave filter was originally designed as a research tool for use in the spectral analysis of sounds.
The amplitude of the output of each filter band can be separately measured and those measurements
plotted on a graph to show the spectral content of the signal. New versions of the spectral analyzer
generate data suitable for feeding directly to a computer for measuring complex time/spectrum
relationships.
The most complex filter system around is the VOCODER. This device contains a filter bank set up for
spectral analysis of a signal and a similar filter bank set up to process a second signal. The measurements
derived from each filter in the analysis bank are used to control the amplitude of the signal from the
corresponding filter in the processing bank. This system will impose the spectral shape of the analyzed
signal onto the processed signal. The effect produced is quite striking, especially if voice is used as the
sound for analysis.
LIMITERS
It is a very common requirement of electronic devices that the amplitude of the input signal must not
exceed some value. The typical result of failure to obey that rule is extreme distortion and possibly
damaged speakers. If the operator of an audio system cannot be trusted to keep the signals within bounds,
it is necessary to include a LIMITER, a device which turns the gain down whenever the signal exceeds the
limit.
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The action of reducing the amplitude of a signal can easily become a form of distortion, so the design of
limiters is a delicate art. The usual approach is to set up the circuit so the transition from free to limited
operation is somewhat gradual, and so that the transition back (called the RELEASE) is even more
gradual. Appropriate release times vary with the nature of the program material, so this parameter is often
adjustable. The limiting point (called the THRESHOLD) is of course also adjustable.
COMPRESSION
It is also possible for the soft spots of a program to become too low in amplitude, especially in a noisy
medium like AM radio. A program with a wide dynamic range may have to have that dynamic range
reduced in order to fit a particular audio system. (Surprisingly, the current switch to digital audio is
making this problem more common as stations try to fit the wide range of compact discs through the
restricted dynamic window of FM transmitters.) A device which reduces the overall dynamic range of a
signal is called a COMPRESSOR.
A compressor works by measuring the average amplitude of the input signal and using the information
from that measurement to control the gain of an amplifier. If the measurement comes out above an
adjustable threshold, the gain is reduced, if the measured input amplitude is below the threshold, the gain
is increased. The amount of effect is usually adjustable and is expressed as a ratio of input dynamic range
to output dynamic range, such as 2 to 1 or 3 to 1.
The release time is adjustable just as with the limiter, but the adjustment is much more critical, because
the compressor is constantly changing gain whereas the limiter only cuts in occasionally. It is impossible
to find a release time that is perfect for all situations. A very short release time will cause the circuit to
attempt to trace the waveform of low frequency components of the signal, resulting in intermodulation
distortion of high frequency components; whereas a long release time will leave the amplifier in the wrong
mode if the signal changes quickly. That second error results in brief bursts of expansion, a very
disconcerting effect. (Sounds like heavy breathing.)
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EXPANSION
Expansion is the opposite of compression. It is accomplished with the same circuit as compression, but the
rules are changed so that any signal with amplitude above the threshold will be amplified and any signal
with amplitude lower than the threshold will be attenuated. Expansion is used primarily for special effects,
such as super loud tacos.
NOISE GATING
Sometimes a program is encountered which has proper dynamic range but which has objectionable
background noise. Very little can be done to reduce the noise without affecting the signal, but the spaces
where there is no signal can often be cleaned up with a NOISE GATE. This circuit is an expander that
only works on the low amplitude parts of the signal. Any section that is below the threshold will be
attenuated rather drastically, so the space becomes nice and quiet. You can hear this device in action on
old movies on TV. Release time adjustment is very critical in this application also, because if the circuit is
too slow, the turning down of the noise will be painfully obvious, but if the circuit works too fast, the
reverb at the ends of the sounds will be lost.
X-Newsgroups: alt.music.4-track,alt.answers,news.answers (not yet)
Subject: 4-track Recording FAQ v0.44 [alt.music.4-track]
Followup-To: alt.music.4-track
X-From: tom@zipcon.net (not yet)
Summary: The frequently asked questions list for alt.music.4-track.
Contains technical information and opinions on hardware, effects,
and recording techniques relevent to 4-track and other
home recorders. Also contains pointers to related online, print,
and video resources, and a brief explanation of some acronyms
and technical terms.
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X-Archive-Name: music/4-track-recording/4-track-faq (not yet)
X-Posting-Frequency: monthly (not quite)
URL: http://homerecording.com/4trackfaq.txt
Maintained-By: David Fiedler
4-track FAQ Version 0.44 DRAFT 99/05/19 17:40
This is a Frequently Asked Questions list for the alt.music.4-track
newsgroup, for music makers who use 4-track recording hardware.
This file may be accessed with the Uniform Resource Locator (URL)
http://homerecording.com/4trackfaq.txt
with world-wide web software (e.g. lynx, mosaic, netscape) or by ftp
at the URL:
ftp://ftp.homerecording.com/pub/record/4trackfaq.txt
NOTICE: Since this is a first stab, it is unstable. It may (should)
undergo drastic revision.
Please contribute questions, answers, and suggestions. Comments
welcome (for the moment). Note that this version of the FAQ probably
has known errors that the maintainer just hasn't had time to fix. The
FAQ maintainer is David Fiedler (mailto:dragon@homerecording.com).
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Warning: Send me any SPAM and I'll make your life a living hell...
CONTRIBUTIONS: Anywhere there are two or more ??s in a row, that
indicates missing information that I would like you, the reader, to
supply. Also, some of the questions are organically growing huge; an
attempt at a concise rewrite of any question would be appreciated. Of
course all other types of contributions are welcome.
**********************************************************************
WARNING WARNING WARNING WARNING WARNING WARNING WARNING
I TAKE NO RESPONSIBILITY FOR THE ACCURACY OF THIS INFORMATION. Sorry.
I'll do my best.
END OF WARNING END OF WARNING END OF WARNING END OF WARNING
Statement adapted from Russ Hersch's FAQ FAQ:
I disclaim everything. The contents of this article might be totally
inaccurate, misguided, or otherwise perverse - except for my name
(hopefully I got that right).
Copyright (c) 1999 by David Fiedler. All rights reserved.
This FAQ may be posted to any USENET newsgroup, on-line service, or BBS
as long as it is posted in its entirety and includes this copyright
statement.
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This FAQ may not be distributed for financial gain.
This FAQ may not be included in commercial collections or compilations
without express permission from the author.
**********************************************************************
To search for a question with a program, look for Q#, e.g. "Q1.1".
List of contributors (knowing or unknowing). If your name is
wrongfully omitted, or you wish it (and your contributions) removed,
email the FAQ maintainer.
adouglas@belvoir.com (Andrew Douglas)
ak748@detroit.freenet.org (Daniel W. Newport)
benl@mojo.europe.dg.com (Ben Last)
billy-em@telalink.net (Billy Center)
brianb@scorpion.iii.net (brian q. buda or Marc?)
bwill@teleport.com (Brad S Williams)
bwm1894@u.washington.edu (Brett McCarron)
cc@hplb.hpl.hp.com (Costas)
cheyenne@netcom.com ("Will" using the account of Melissa Duncan)
crick@ccnet.com (TEFKAR)
dfrankow@winternet.com (Dan Frankowski)
davidc@access.rrinc.com.blacksburg.va.us (David Copeland)
dragon@homerecording.com (David Fiedler) (Dragon)
ertrinid@girtab.usc.edu (Elson R. Trinidad)
gajoob@utw.com (by way of gajoob@utw.com (Bryan F. Baker))
haibachi@tiamat.umd.umich.edu (Jay Itchon)
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jeibisch@revolver.demon.co.uk (James Eibisch)
jmireau@tibalt.supernet.ab.ca (James Mireau)
kowalski@scooter.ping.de (Thorsten Kowalski)
ks43@cornell.edu
leiter@panix.com (Phil Dahl)
lwillia@ix.netcom.com (Larry Williams)
macchi@marina.scn.de (Gian Carlo Macchi)
marcl508@hudson.iii.net (Marc LaFleur)
mcknight@pire.org (Scott McKnight)
michaelj@hstl.ultranet.com (Michael J. Slaney)
mparrott@kendaco.telebyte.com (Michael Parrott)
mucilage@unity.ncsu.edu (Ross A Grady)
mwdrews@mailbox.syr.edu (Mark Drews)
nigelsp@rain.org (Nigel Spencer)
nflorin@med.unc.edu (Nathaniel Paul Florin)
pseo@mail2.sas.upenn.edu (Peter S Seo)
psu02178@odin.cc.pdx.edu (Joshua Meredith)
me@ram.org (Ram Samudrala)
reedijk@gene04.med.utoronto.ca (Rob Reedijk)
robertb@primenet.com (Robert Blackwell)
rwh8234@erc.jscc.cc.tn.us (Robert W. Hough)
schmange@wbb.com
shitcan@eskimo.com (Michael Crowl)
sratte@mindvox.phantom.com (Swamp Ratte)
stabnste@phoenix.phoenix.net (Erik Karlson)
talarczyk@ACFcluster.NYU.EDU (Michael J. Talarczyk)
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taroh@kohnolab.dnj.ynu.ac.jp (Dr. Taroh SASAKI)
tmbs@mailserv.mta.ca (monkey 100)
tnolan@frymulti.com (Timothy M. Nolan)
toehser@cais2.cais.com (Tom Oehser)
tstrohma@theodolite.ae.calpoly.edu (Trevor Strohman)
will@iglou.iglou.com (William M. Willis)
zz93f010@polar.etsiig.uniovi.es (Perez Sanchez, Florian Manuel)
Changes between 0.43 and 0.44:
------------------------------
various spelling and grammar errors
Preamp definition added to
clarification of 4-track mixdown
updated table for 424s
Changes between 0.42 and 0.43:
----------------------------------
Maintenance taken over by David Fiedler
(previously Maintained-By: Dan Frankowski who did a great job!!)
URLs and other minor corrections
Some changes between 0.41 and 0.42:
----------------------------------
- added ref to a book by Bruce Bartlett.
- added headers in anticipation of being listed in news.answers someday.
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Some changes between 0.4 and 0.41:
----------------------------------
- added ref to the awesome rec.audio.pro FAQ.
- major new info in 4-track comparison table! See Q1.3.
- balanced and unbalanced ins and outs
- rewrite of def of EQ
- difference between amp and pre-amp
- another opinion on "how to get a good sound", Q3.1.
Some changes between 0.31 and 0.4:
----------------------------------
- Added ref to IMPROVIJAZZATION NATION shamelessly lifted from GAJOOB DiY
- Added ref to MIDI book
- Added refs to rec.music.makers.songwriting and alt.music.lyrics
- Added ref to copyright FAQ
- Added questions on other computer software, answers not given.
- Expanded discussion of computers and home recording, including
SAW software, and Macintoshes
- Edited definition of EQ and noise reduction
- Added ref to Audio Technica RMX 64 4-track machine
- Added ref to DAT-heads FAQs and email list
- Added ref to the Buddy project
- Added reviews of two books and a great videotape
- Added definitions of DiY, indie, decibels
- Added MTC definition
- Added SMPTE definition and discussion
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- Added explanation of gain and volume, and the difference
- Added ref to Sound Effects and Digitrax software
- Added discussion of using compression
- Added line level voltages
- Added discussion of mics
- Added ref to my SAW ftp site
- Added discussion of using Macintoshes
- Added GAJOOB ref
- Added 8-track comments
- Added discussion of improving drum sounds
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Table of contents
-----------------
Q0. Philosophy
S0. Definitions and Acronyms not covered elsewhere.
Q0.1 What is MIDI?
Q0.2 What is MTC?
Q0.3 What is SMPTE?
Q0.4 What is gain? What is volume? What is the difference?
Q0.5 What is "DiY"? What is "indie"?
Q0.6 What are decibels? When are they used instead of volts?
S1. Hardware
Q1.1 What is a 4-track machine?
Q1.1.1 What is an FX (effects) loop?
Q1.1.2 What is EQ (ee-kyoo)? What kinds are there?
Q1.1.3 What is noise reduction? What is Dolby? What is DBX?
Q1.1.4 What are balanced and unbalanced ins and outs?
Q1.1.5 What is the difference between a pre-amp and an amp?
Q1.2 What should I look for in buying a 4-track machine?
Q1.3 What are some brands of 4-track machines?
Q1.3.1 What are some good brands of 4-track machines?
Q1.3.2 What is the difference between a Tascam 424 and 464?
Q1.4 How important are microphones?
Q1.5 What are some brands of microphones?
Q1.6 What is a "phantom powered" microphone?
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Q1.7 Can I use a personal computer to digitally record and mix music?
Q1.7.1 How is Turtle Beach's Quad Studio package?
Q1.7.2 How can I get a demo of SAW (Software Audio Workshop) software?
Q1.7.3 What is a MIDI sync device?
Q1.7.3.1 What is an SMPTE-to-MIDI converter?
Q1.7.4 What is Sounds Effects?
Q1.7.5 What is DigiTrax?
Q1.7.6 What is Cakewalk?
Q1.8 How does an 8-track machine compare to a 4-track machine?
Q1.9 How should I clean my 4-track machine?
Q1.9.1 Do I need to demagnetize my 4-track heads?
S2. Effects
Q2.1 What is flange?
Q2.2 What is reverb?
Q2.3 What is compression?
Q2.3.1 Do I need stereo compression?
Q2.3.2 Should I use compression on drums? Vocals? Bass?
Q2.4 What is limiting? How is it different from compression?
Q2.5 What is (upward) expansion?
Q2.6 What is companding?
Q2.7 What is a noise gate?
Q2.8 What are some brands of effects boxes?
Q2.9 Should I get an all-in-one effects box or a more basic box and
a separate compressor/limiter?
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S3. Recording Techniques
Q3.1 How do I get a good sound?
Q3.2 What should I know about speeding up recordings?
Q3.3 What should I know about using a VCR as a digital recorder?
Q3.4 What should I know about doing an external pre-mix versus
an internal bounce?
Q3.5 Can I use compression on a mixed signal?
Q3.5.1 Is it better to compress when recording or mixing?
Q3.6 How do I improve my drum sound?
Q3.6.1 How would I use noise gates to improve my drum sound?
S4. Related information
S4.1. Information online
Q4.1.1 What are some related newsgroups?
Q4.1.2 What are some related FAQ lists?
Q4.1.3 What are some related email lists?
Q4.1.4 What are some related WWW (World-Wide Web) pages?
Q4.1.5 What are some related ftp sites?
S4.2. Information not online
Q4.2.1 What are some related magazines?
Q4.2.2 What are some related books?
Q4.2.3 What are some related videos?
S4.3. Company information
Q4.3.1 What is Tascam's address and phone #?
Q4.3.2 What is Musician's Friend?
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--------------------------------------------------
Q0. Philosophy
A short quote from an alt.music.4-track reader:
From mucilage@unity.ncsu.edu Thu Sep 15 02:29:08 1994
Message-ID:
"Now, of course, I'm not Mr. Hi-fi, but then any idiot who goes crazy
trying to replicate perfect hi-fi sound in a home studio is
by-and-large nuts, when he/she could just save up that money and go
rent studio time in a 16-track analog studio with a decent mic package
for about $30 - $35 an hour.
"I've got some advice fer you: spend a little more time writing songs and
practicing them, and a little less time tweaking your superbedroomstudio.
I've heard more clean-sounding 4-track demos of more shitty songs than
I'll ever care to elaborate on.
"And always remember this: every single device in the signal path alters
the signal. Period. There is no perfect transparent mic. There are not
even different levels of alteration, really--just different types, some
of which sound more "normal" than others.
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"So your best bet is to use whatever equipment makes the coolest
sound--whether it's a zillion-dollar vocal mic or a paper cup."
S0. Definitions and Acronyms not covered elsewhere.
Q0.1 What is MIDI?
Musical Instrument Digital Interface. A standard which allows sending
and receiving of messages such as "Play an A 440 for 100 ms with patch
#9 and some pitch bend." Since many instruments and controllers use
this standard, you can hook up a lot of equipment.
There are many books on this subject. Go to the library. See also
Q4.2.2.
Q0.2 What is MTC?
MIDI Time Code.
Q0.3 What is SMPTE?
Adapted from talarczyk@ACFcluster.NYU.EDU (Michael J. Talarczyk):
Society of Motion Picture and Televison Engineers (time code).
See also Q1.7.3.
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Q0.4 What is gain? What is volume? What is the difference?
Adapted from tstrohma@theodolite.ae.calpoly.edu (Trevor Strohman):
Gain occurs before the preamplifier stage, volume occurs afterward.
When to fiddle with each:
Adjust gain levels once, to find the optimal input levels for your
mixer/4-track: high enough for a good signal-to-noise ratio, low
enough for the desired level of distortion (usually none). Then
adjust volume controls to change the levels in the mix.
Lowering the gain reduces clipping and distortion. The gain also
helps even out levels: you don't have to have one volume slider
cranked while another is almost off to get a good mix. However, the
mix happens in the volume controls.
Cranking gain too high is much more likely to cause distortion than
cranking volume, and more noise occurs on low gain than on low volume.
Q0.5 What is "DiY"? What is "indie"?
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"DiY" - Do-it-yourself (music)
"indie" - Independent, meaning not released on a major record label.
Q0.6 What are decibels? When are they used instead of volts?
From a Webster's online dictionary:
deci-bel \'des-e-,bel, -bel\ n
[ISV deci- + bel]
(1928)
1a: a unit for expressing the ratio of two amounts of electric or acoustic
signal power equal to 10 times the common logarithm of this
ratio
1b: a unit for expressing the ratio of the magnitudes of two electric
voltages or currents or analogous acoustic quantities equal to
20 times the common logarithm of the voltage or current ratio
2: a unit for expressing the relative intensity of sounds on a scale
from zero for the average least perceptible sound to about 130 for
the average pain level
3: degree of loudness; also: extremely loud sound -- usu. used in
pl.
From the DAT-heads microphone FAQ:
"Noise is typically referred to in microphones in terms of equivalent
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sound pressure level.. The measure used is typically dBA: decibels
above the hearing threshhold of 0.0002 microbar, A-weighted."
tstrohma@theodolite.ae.calpoly.edu (Trevor Strohman) writes:
A decibel is a logarithmic power level. Since mics output such tiny
signals in comparison to line levels, people use dB to talk about
levels. I'm not quite sure what all the levels are in relation to.
However, the formula for decibels is:
/ Vin \
dB = 10*log | ----- |
10 \ C /
Where Vin = the voltage you're measuring and C is the voltage you're
going to call 0 dB. I couldn't tell you in this case what C is, but
I'm assuming it's somewhere around 1 volt.
dB make the numbers more manageable. I can say "It's at -30 dB", or I
can say that "It's at one thousandth the voltage." Some mixers claim
a dynamic range of up to 90 dB or more! I'd rather say -90dB than one
billionth the voltage. I don't usually have reason to talk about
voltage of audio signals much.
S1. Hardware
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Q1.1 What is a 4-track machine?
A 4-track machine is an audio device that allows the user to record
four separate channels, perhaps simultaneously or perhaps at different
times (possibly out of sequence), so that they may be played back
simultaneously.
The ability to record at different times or out of sequence
distinguishes 4-tracking from recording onto a stereo deck.
Q1.1.1 What is an FX (effects) loop?
'FX' is the usual abbreviation for 'effects' in scripts and stage
directions.
An effect loop 'splits off' the signals on their way out of the
4-track (or mixer) and sends them to an external effects unit. The
output of that unit is then fed back into the output of the 4-track /
mixer. You can set, for each track, what level of signal is sent to
the effect unit, so you can have some tracks heavily treated and some
not treated at all.
For example, suppose you have 4 tracks of soaring music on tape,
including a trumpet. If you played the whole mix through a reverb,
it'd be like listening to it in a cathedral; unusably reverbed. Thus,
you want to add reverb to the trumpet.
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The effect unit is set to produce a 'wet' signal because that's what
you want to feed back into the final signal from the 4-track or mixer.
Remember that this is being added to the original, untreated signal,
so there's no point in adding, for example, partially reverbed trumpet
to unreverbed trumpet. What you do is to choose the level of the
trumpet track which is sent to the reverb so that the right level of
reverbed trumpet is heard in the final mix.
Q1.1.2 What is EQ (Ee-Kyoo)? What kinds are there?
Thanks to Michael Parrott for a nearly
complete rewrite for this question.
EQ (where each letter is pronounced: Ee-Kyoo) stands for
"equalization." As a noun, it means those controls on a recording,
mixing, or playback unit which allow for altering the tonal
characteristics of an audio signal by boosting (increasing) or cutting
(decreasing) the prominence of specific frequencies or frequency
ranges within the signal. As a verb, it indicates use of those
controls.
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Michael Parrott writes:
"Note that EQ does not tend to make extreme changes in signal levels;
instead, it can be used to reduce or increase the prominence of
certain frequencies in the signal, which in turn reduces or increases
the audible prominence of certain portions of a track or mix. If you
get a chance to experiment, listen to what happens when you boost the
5 KHz range in an evenly-mixed guitar track; the guitar should become
more prominent in the mix without having significantly changed it's
signal level."
Different types of EQ:
(1) Parametric EQ (also "sweepable EQ")
A form of EQ which affects broad sections, or bands, of frequencies in
an audio signal. May be found as either two controls (Low and High),
three controls (Low, Mid, and High), or four controls (Low, Mid, Mid
Freq, and High). They tend to affect frequencies in the following
ways:
Low: Boosts or cuts frequencies in the low (bass) half of the audible
frequency spectrum (20 Hz - 1 KHz). A "shelving" control, it tends to
make more extreme changes at very low frequencies (20 Hz) and less
extreme changes at higher frequencies (1 KHz).
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Mid: Boosts or cuts frequencies in the mid range of the frequency
spectrum (100 Hz - 10 Khz). A "peaking" control, it tends to make more
extreme changes at the mid-range frequencies (around 1 KHz) and less
extreme changes at either end of the mid range (100 Hz and 10 KHz).
Mid Freq: Also seen as "Sweep", "Para EQ" and other variations. Alters
the center frequency or "peak" of the Mid EQ control. This allows the
Mid EQ control more flexibility in boosting or cutting frequencies
toward the lower-mid and upper-mid range of the spectrum.
High: Boosts or cuts frequencies in the high (treble) half of the
frequency spectrum (1 KHz - 20 KHz). A "shelving" control, it tends to
make more extreme changes at higher frequencies (20 KHz) and less
extreme changes at lower frequencies (1 KHz).
(2) Graphic EQ
A form of EQ which is generally designed to alter specific, very
narrow frequency bands in an audio signal. The number of controls may
vary from as few as three (effectively another form of parametric EQ)
to as many as 30 or more per channel, with the audible frequency
spectrum evenly divided among them. Controls are usually sliders, but
may also be seen as "plus-or-minus" buttons with an LED or flourescent
display indicating the amount of boost or cut for each frequency.
These controls are of the "peaking" type, centering on a specific
frequency with a small amount of overlap with adjacent controls.
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The term "graphic" is used to describe this type of EQ due to the fact
that the sliders (or other indicators), when set to most people's
listening preferences, tend to look like a sine wave or gentle curve.
Hence, a "graphic" representation of the EQ being applied to the
signal.
This type of EQ allows more precise control than parametric EQ over
the tonal characteristics of a signal and makes singling out specific
frequencies for boosting or cutting much simpler and more effective.
Different ways EQ is applied to the mix:
(a) per channel
Each input channel may be separately EQ-ed, as in a mixer.
(b) per bus
Each bus may be EQ-ed, where a "bus" is one of multiple destinations
for an output signal. Example buses: the master mix, feedback
monitors for the musicians, or the effects (FX) loop.
(c) global
The whole output sound may be EQ-ed. The tone controls on a hifi
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amplifier are, therefore, global EQ.
Q1.1.3 What is noise reduction? What is Dolby? What is DBX?
See the rec.audio.pro FAQ for incredible detail about this.
There are two major families of noise reduction technology which
you're likely to find on 4-tracks: Dolby and DBX. These are each
discussed in a paragraph below.
There are (at least) three varieties of Dolby on cassette decks, but
they all work more or less the same way: whilst recording, they
enhance the high frequencies in the same area where hiss occurs.
During playback, they reduce those frequencies back to the same level
they were originally. This also reduces the hiss. You can play back
a Dolby-encoded tape without Dolby; the only effect you'll hear will
be a brighter, higher top end.
DBX processes the sound more severely than Dolby [and more
effectively?? --DSF]. If you record using DBX, you *must* play back
with it on. You may even find that a DBX-encoded tape from one
machine does not replay exactly on another.
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Q1.1.4 What are balanced and unbalanced ins and outs?
reedijk@gene04.med.utoronto.ca writes:
An unbalanced signal pathway is identified by having two wires (RCA
plugs, guitar jacks, some mic cables). These are high impedence and
are less desirable because as the cable gets longer, more noise is
introduced into the signal. A balanced signal pathway has three wires
which is most common in mics (SM58s for example). A few 4 tracks
have balanced inputs whcih means you can use better quality mics
without adaptors. AT RMX64s as far as I know are the only ones with
balanced outputs which you would use for hooking it up to high-end
gear (I have yet to use them!)
Q1.1.5 What is the difference between a pre-amp and an amp?
reedijk@gene04.med.utoronto.ca writes:
Pre-amps ("pre-amplifiers") shape the character of the signal. This
is where the equalization and gain (sometimes distortion) are
controlled. An amp simply generates the raw power to push a speaker.
A bad pre-amp can distort the signal in many ways including clipping,
compression, poor frequency response (e.g. the "highs" disappear or
sound harsh etc.) or noise.
dragon@homerecording.com adds:
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Preamps are designed to boost a mic level signal to a line level also.
Q1.2 What should I look for in buying a 4-track machine?
Adapted from correspondence with Ben Last :
First, understand how 4-tracks work and how you would use the unit
you're thinking of buying in a real, working, setup. Consider how
you'd record and mix a track with it, from 'which instrument plugs in
where' to 'how many bounces will I need' to 'how will I eq the final
mix.' If you can't answer these questions, you're not going to get
something *wrong*; it's just that the more you look into it, the more
you will (hopefully) understand about what the various systems on the
market can do.
Below are some specific questions to ask:
(1) How many simultaneous tracks can I record?
Most modern 4-tracks will handle 4 at once; lower end systems only
operate with a single stereo bus, and can therefore handle only two at
once. Consider how many people you might have playing at once.
For example, I work with one other guy (I suppose two is too small a
number to constitute a 'band' :-); both usually play guitar. Thus,
since the synth backing (all the MIDI) comes in as two tracks
(stereo), we need to be able to record 4 tracks as well so that the
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two guitars each get a track of their own.
(2) Does the 4-track machine run at double speed?
This is probably almost mandatory in a quality 4-track these days.
However, if you want to play normal cassettes in it, or record for
replay in a normal deck, then you need the ability to run at normal
speed. Note: it is *not* normal procedure to record a tape on a 4-track
for replaying in a normal cassette deck, since the 3rd and 4th tracks will
come out backwards. You must mix down to an external machine instead.
(3) Does the 4-track machine have appropriate noise reduction?
If at any time you want to record onto a cassette to be played back on
a normal deck, you need to consider that normal decks will not have
DBX, only Dolby, so you either use no noise reduction, or Dolby.
See also Q1.1.3.
(4) Is noise reduction controllable per track while recording?
If you are using MIDI and tape sync you need to ensure that you can
record sync with no noise reduction on one track (usually 4) and get
it back out without hearing it in the mix. Usually track 4 can have
the noise reduction switched on or off separately.
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(5) Does it have an outboard FX (effects) loop; can you use it to do
most (if not all) of your mixing, or will you need an outboard mixer?
See Q1.1.1 for a discussion of an FX loop.
(6) If you're mixing with it, what eq does it have?
Most 4-tracks will have some form of eq; it may be per channel or a
stereo graphic that covers all the tracks. If you're trying to keep
one instrument per track, will the eq allow you to affect just that
one track? If eq is a real issue, you need to look at some sort of
external mixer.
(7) Will the inputs switch from mic to line level?
Signals are classified into (roughly) two sorts, based on the voltages
involved: 'mic level' signals are lower voltage, typically seen from
microphones (of course) and electric guitars; 'Line level' signals are
higher voltage, typically seen from synths, hi-fis, etc.
NOTE: Guitars can provide a hell of a signal if played heavily,
especially Les Pauls!
Most 4-tracks will at least have a 'line / mic' switch on the input.
Some have a slider to allow the best matching to be selected.
It's important to get this right: you can connect a mic level signal
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to a line level input and hear the sound by turning the volume up, but
you'll also turn up the noise. Similarly, you can put a line level
signal into a mic level input and turn the channel down, but you may
overload the input and cause distortion.
Typically, you want to see a good strong (but not distorting) signal
when the input channel is a 80% or so of maximum volume. With a
slider to set the input impedance, you can get the best 'match'
between instrument and 4-track.
tstrohma@theodolite.ae.calpoly.edu (Trevor Strohman) writes:
Line levels come in two types: -10 dB and +4dB. Consumer equipment is
usually -10dB and pro equipment is -10 or +4. Usually the difference
between the two types is small enough that the gain control on the
mixer/4-track can handle the difference.
For a 4-track studio, +4 signals almost are never come across.
Examples of consumer line level equipment include tape decks, CD
players, and synthesizers. Record players are _not_ line level and
require a preamp. Their voltages are comparable to microphone levels.
Electric guitars are tricky, because they are usually stronger in
voltage than a microphone and yet lower than a line-level signal. Use
the gain knob wisely.
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Typical voltages for these signals are roughly 1 volt for line level
signals, and maybe 100 millivolts for a microphone (depends on the
microphone type).
Remember to give yourself plenty of room to mix. Often I see people
with the volume sliders all the way up. This gives no flexibility.
If you have input gain knobs, play/sing at a normal level through all
sources and adjust so that there's equal volume coming out at equal
slider positions, and so that you don't have to push the slider too
high to get a good amount of volume. Now you have total flexibility
to mix with.
Q1.3 What are some brands of 4-track machines?
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Table of Features
Note: this is way out of date. If you have updated info,
please email it to dragon@homerecording.com in the same format it is in
the table, i.e. just a line like:
Phlegm X007 666 Y N N Y N 9 4 Y 3P Y Y Y Y
This info is culled from different sources and may be WILDLY INACCURATE!
Brand Price(US$) NS DS SS NR #I #X #R FX EQ ST SY PT PI
--------------------------------------------------------------------------
Audio Technica
RM X64 ??? Y Y N C 6 -- -- 2 P Y N -- --
Fostex 280 949 (699) -- Y -- D 8 -- 4 Y ?P Y -- -- Y
Fostex 380S 1195 -- Y -- D 6 -- 4 Y 3P Y -- Y Y
Fostex X-26 ??? Y N N B 6 -- 2 Y 3P Y? Y -- --
Fostex X18 350-500 (new) Y N N B 4% -- -- @ -- Y Y -- --
Fostex X18-h 450 -- -- -- -- -- -- -- -- -- -- -- -- --
Fostex XR-3 399 Y N N D 2 -- 2 -- ?P -- -- Y --
Fostex XR-5 499 N Y N D 4 -- 2 Y 2P Y Y Y Y
Fostex XR-7 699 Y Y N D 6 -- 4 Y 3P Y Y Y Y
Marantz PMD740 999 Y Y N -- -- -- -- Y 3P Y Y -- Y
Tascam Porta 01 350 (used)
Tascam Porta 03 150 (used) Y N N D 2 -- 2 -- 2P -- -- -- --
Tascam Porta 07 499 (400) N Y N D 4 -- 2 Y 2P Y -- -- --
Tascam 424 250 (used) Y Y Y X 4 -- 4 Y 2P -- Y Y --
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Tascam 424Mk2 300 Y Y Y X 6M2S 2 4 2 3S Y Y Y --
Tascam 424Mk3 479 (600) Y Y N X 6M2S 2 4 2 3S Y Y Y --
Tascam 464 999 (800) Y Y N X 8* -- -- Y 4+ Y Y -- --
Tascam 644 1599 (1100) Y Y N D 16 -- 4 Y ?P Y Y -- Y
Yamaha MT120 -- Y Y -- X 4 -- -- Y G Y Y -- --
Yamaha MT120S 579 (list) Y Y -- D 4 -- 4 Y ?G Y Y -- --
Yamaha MT4X 599 (540) Y Y N D 4 -- 4 Y 3P Y Y Y Y
Yamaha MT50 449 (400) Y -- N D 4 -- 4 Y 2P Y Y -- --
* - 4 of these are stereo inputs.
+ - 4 channels have one high, one low, and two mid. The others have
just one high and one low.
% - it also has an aux return.
@ - you have to use the mono-mix output as an effects send and the aux
return as an effects return during mixdown
(Note: Any block containing "??" or "--" means I don't have that
information for that unit; if you happen know, please let me know!)
Features in table above:
Brand - The company and name of the machine. Listed in alphabetical
order.
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Price - Price is in U.S. dollars, manufactuere's list unless otherwise
specified. Sorry international folk. Stores will often give a
discount from the list price. For example, a store offered me a
Tascam Porta 07 for $400 (list $499). Prices in parenthesis indicate
approximate 'street' prices of a fairly new unit (probably inaccurate
and surely highly dependent on location and condition of the unit).
NS - Normal speed record/play (4.8 cm/s or 1.875 in/s). This allows tapes
to be recorded at the speed of a normal cassette deck.
DS - Double speed record/play (9.5 cm/s or 3.75 in/s). This allots
double the tape to the same amount of time as normal speed, thus
improving sound quality.
SS - Half speed play (2.4 cm/s or .9375 in/s). This allows you to
play quick phrases slowly so you can hear them.
NR - noise reduction. D for Dolby (B for Dolby B, C for Dolby C if known),
X for DBX.
#I - Number of simultaneous inputs (mono and/or stereo).
#X - Number of XLR inputs
#R - Number of simultaneously recordable tracks.
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FX - Has an effects loop or aux send/return buss.
EQ - Type of equalization on mic inputs. First character is number
of controls, second character is type: "P" for parametric,
"G" for graphic, "S" for swept mids.
ST - Stereo out channel or buss.
SY - MIDI tape sync capability.
PT - Pitch control.
PI - Punch In/Punch Out editing.
#X - Number of XLR inputs
Q1.3.1 What are some good brands of 4-track machines?
Ben Last recommends the Yamaha MT120 ("'cos I bought mine after much
research!"). Double / normal speed, DBX noise reduction, 4
simultaneous inputs, FX loop, graphic eq, monitor / mix / stereo out,
tape sync capability. 320 pounds (UK) in 1992. They are no longer
sold new.
reedijk@gene04.med.utoronto.ca (Rob Reedijk) writes:
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There is a brand missing from your info: Audio Technica RMX 64. I
have not one but two and I think they're fantastic.. They only made
them for short period (around '85) and I think they stopped because
they cost too much to manufacture. They have 6 full channels plus two
returns which can also be used as channels, parametric e.q., balance
and unbalance inputs on all six channels, balanced and unbalanced outs,
phantom power on all six channels, Dolby B and C, two speed with pitch
adjust, real V.U. meters, zillions of inputs and outputs on the back,
4 headphone outs etc, etc. They also weigh 50 lbs. Because in
reality they contain a sort of pro mixer they sound fantastic and
they're built like tanks. They are kind of low tech in fact that
there is no auto locate programmable punch-in type features. Get one
if you can. I bought my first one in 1985. I bought my second one
last year for $225 Canadian (about 4 cents U.S.). Compare that to a
new fostex $1000+ made of plastic planned obsolescence disaster
(apologies to the fostex contingent).
The reason I have two is I can connect them together and get 10
channels (two used for signal routing). I can also bounce 4 to 2 with
only one generation of signal loss. Furthermore, I have, this way, an
excellent stereo cassette directly linked up for final mixing.
ram@indigo3.carb.nist.gov (Ram Samudrala) writes:
Needless to say, the Tascam 464 is one of the best analog 4-tracks out
there, IMO. It is particularly useful if you're a solo recording
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artist and want the computer automation to do a lot of the punching
in, etc. for you. I have found the mixer to be quite superior, and I
think the noise reduction system works quite well at high speed.
Q1.3.2 What is the difference between a Tascam 424 and 464?
toehser@cais.cais.com (Tom Oehser) writes:
>What does the 464 have that makes it sell for $230 more than the 424?
>(Best I can tell, the difference is extra channels in the mixer and XLR
>inputs. Is this difference worth it?)
More channels. XLR and preamps. I think the 424 has a wall-wart and
the 464 has an internal power suppoly. Computerized auto-locate and
auto-punch. Large LCD display with 6 meters, 424 has LED with 4
meters. More/better EQ and cueing section controls. Just overall,
more bells and whistles. Yes, I think it is worth it, unless you have
a good mixer with preamps, like a Mackie 1202, and are willing to
forego the extra automation.
Q1.4 How important are microphones?
You can't fix a bad sound in the mix. Start with a good source (a
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good mic) and you're halfway there.
Adapted from tstrohma@theodolite.ae.calpoly.edu (Trevor Strohman):
It happens all too often. A group is looking for great sound on a
limited budget. They go to the music store and see all the glitzy
4-tracks and other recorders. Maybe they have deeper pockets and go
straight for an ADAT. Then they need a microphone. Microphones, not
having the sizzle factor of the recorders, are last on the list, and
there's only $50 left. They leave with a digital recording studio and
a Radio Shack tie-clip microphone.
They get home and plug it in. The features work just like they did in
the store, but the sound isn't quite right. So they go back with more
money. They still skip the mics, and go for compressors, equalizers,
reverbs. They have a little money left over. This time they buy a
Shure SM57.
They go back and try it out. Neat sounds! Not quite like the CDs,
but it sounds pretty good. They want more. They go back to the
store... the cycle continues.
Of course, there will always be this cycle with musicians looking for
the perfect sound, but you can get to the perfect sound a lot faster
if you invest in microphones first. As they say in computing, garbage
in, garbage out. No effects can compensate for a terrible microphone.
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When you're at the store looking at boxes, think about it. Do you
really need to spring an extra $300 for a half-octave equalizer, or
could you spend the same $300 on a very good home mic and not need the
EQ at all? This works for speakers too. My church runs the most
horrible sounding monitor speakers (12" woofers in little boxes) and
uses a very expensive equalizer to try to clean up the sound. The
sound that comes out still sounds terrible (not as bad, though), but
if they'd just taken the money and invested in new speakers, they
could have much better sound.
Of course, you need to do what's right for you, but I would say to at
the very least spend 50% of what you do on the 4-track on microphones.
I would try for 100%. Check the DAT-Heads microphone FAQ before
buying.
Q1.5 What are some brands of microphones?
[I'll leave the previous text in for a few versions, but all of this
is probably superseded by the impressive-looking DAT-heads microphone
FAQ. See Q4.1.2. --DSF]
Realistic (i.e. Radio Shack) omnidirectional US$11.99
nflorin@med.unc.edu (Nathaniel Paul Florin) writes:
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"The $11.99 omni-directional ones (don't know what they're called;
they come in a clear pack that hangs by a peg in the store and come
with a little stand) work very well for electric guitars."
"Replacement" US$23.99
nflorin@med.unc.edu (Nathaniel Paul Florin) writes:
"an electret condenser mic; it comes in a little red box with a stand
and windscreen and needs a new AAA battery every six months. Works
great for vocals, acoustic guitars, banjos, ukeleles, and for light
percussion."
Shure SM-57 US$99
Shure SM-58 US$109
bwm1894@u.washington.edu (Brett McCarron) writes:
"A '58 will work well for vocals, Will also add a warm sound to mic'ed
guitar amps. I used to use an SM58 when mic'ing my Marshall stack.
Most people would opt for the more universal Shure SM57, though."
Shure PROLOGUE US$25-30
Shure SM-81 US$279-300
Electrovoice microphone (ND 357) US$150
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am@indigo3.carb.nist.gov writes:
"[The Electrovoice] is better than any of the Shure microphones I tried
out. The clarity of the sound and the way it captures the reverb in a
room is amazing. It is very crisp, and great for recording the flute
or acoustic guitar."
Other brands:
Neumann U-87 ~US$1200
Audio-Technica
Q1.6 What is a "phantom powered" microphone?
benl@mojo.europe.dg.com (Ben Last) writes:
A microphone that is supplied with 48V (usually) via the same cable
that connects to the mixer / whatever. It's unusual to find any
4-track gear that provides phantom power; it requires a balanced line
(3 wire) cable. You can get external boxes to phantom power such a
mic, which will also give you the micx signal on a standard 1/4" jack.
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ak748@detroit.freenet.org (Daniel W. Newport) writes:
Condenser mics, need to be powered (48v) from mic preamp input.
Generally more expensive, more responsive, studio quality mic.
Sometimes can be damaged by high sound pressure levels.
Q1.7 Can I use a personal computer to digitally record and mix music?
mcknight@pire.org (Scott McKnight) writes:
>I haven't really thought too much about digital recording to PCs
>because I assumed that the cost of a powerful enough PC with
>sufficient HD space would be out of my price range. What would be
>involved in setting up the kind of system you're describing from
>scratch? Am I understanding correctly that you're talking recording
>live audio to a PC's HD, not MIDI? What would be involved in getting
>it to lock to MIDI? How about locking to an external analog tape deck?
nigelsp@rain.org (Nigel Spencer) writes:
Yeah we're talking 44.1 khz digital recording here so yeah you need
some reasonable hard disk space. Here's a sample system presuming you
chase down parts from vendors at shows or discount outlets.
Motherboard DX33 minimum - DX 2- 66 US$150-US$300 (VLBUS + DX-2 66 is
cheap now), Case, floppy, VGA card, IDE controller with
serial\parallel IO, US$150 all up
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IDE 420 mb hard disk (SCSI's better but IDE is fast enough & cheap)
US$240 Western Digital, Conner or Quantum (best) can be had for this
price.
8 mb RAM US$320
The monitor of choice is up to you !! whatever you want to spend
So a basic DX33 system would cost you about US$900 + a monitor
obviously. Sure it's not cheap but provides you a platform for
sequencing, patch editing and digital audio, and being a PC can be
upgraded by you as your needs expand, much easier & cheaper than the
MAC. You may see cheap off the shelf systems but usually only have
smallish hard drives and 4 mb of RAM instead of 8.
Quad will generate MTC, that's about all I can tell you, so you can at
least sync an external sequencer. I don't know whether it can chase
lock to tape or not ...??? Ring Turtle Beach and ask them. If you want
full recording power there is SAW which is US$599 software only, but is
phenomenal. Can chase lock, 4 STEREO track simultaneous, automated
faders & mutes, and fast non destructive editing, edit lists, and
automatic crossfade tools for 'razor' edits.
Quad is now available mail order for .... US$350 card & software. If you
want upgrade later to SAW (when you're rich) the Tahiti card that
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comes with Quad is fully supported by SAW.
tstrohma@theodolite.ae.calpoly.edu (Trevor Strohman) writes:
The new Power Macintosh computers and the older 660AV and 840AV Macs
all have 16-bit stereo input and output built-in. The public domain
program Sound Effects is incredible and supports multiple tracks, but
only works when all of the sound is in memory. DigiTrax is around
$200 and supports up to 6 tracks.
The Macintosh has long had a large foothold in the audio market.
ProTools is a major standard but costs $8000 for a decent setup. The
high costs of manufaturing NuBus cards and the few low-end Macs that
support NuBus have kept out the low-end audio market. By fall of
1995, however, new low-cost PCI based Macintoshes will be out (the
high-end PowerMac 9500 PCI has been released), and audio options for
the Mac should increase dramatically as PC PCI audio boards become
Mac-compatible.
Q1.7.1 How is Turtle Beach's Quad Studio package?
Information in March 1995:
ks43@cornell.edu writes:
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I recently found out about Turtle Beach's Quad Studio package. It's
an expansion card that you use with your computer as a digital
four-track recorder. The great part about it is that is only about
US$350. Can anyone give me some feedback on this product? Has anyone
tried it? Thanks.
billy-em@telalink.net writes:
From everything I have seen this is a technology that still needs
time. I was really interested in this product until I logged onto
Turtle Beaches BBS and saw numerous messages from
very upset users about problems with Quad Studio. Have seen several
messages here about people getting it, and sending it right back with
days. It uses about 25Mb per song on average, so you need a Healthy
Disk Drive.
Q1.7.2 How can I get a demo of SAW (Software Audio Workshop) software?
The company which produces SAW is Innovative Quality Software, Las
Vegas, NV, phone 702-435-9077
Someone emailed me a copy downloaded from their BBS; you can find it
BinHex-ed at:
ftp://ftp.winternet.com/users/dfrankow/saw
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or
ftp://ftp.winternet.com/users/dfrankow/saw.gz
-----
billy-em@telalink.net writes:
Another product you might want to check into would be SAW (Software
Audio Studio) you can get a demo from
ftp.vortex.com/audio/SAW
The thing that turned me against this product is that to get around
Bus contentions and other PC and/or Hard drive issues you have to have
an External Midi Sync.
rwh8234@erc.jscc.cc.tn.us (Robert W. Hough) writes:
you can download the SAW demo from their bbs.
[what is the name and phone# of the BBS?? --DSF]
: Also, what is the limitation that the demo holds that the full product
: doesn't?
they say you can only record for 1 or 2 minutes with the demo
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: Finally, how much is the full product and where can you buy it?
any software vendor...i imagine....i'll tell you this, though, the
advertisement is VERY misleading. I have a friend who bought SAW. He
had all the requirments of the ad, but come to find out that he could
only record on 2 tracks. Not enough memory or something.
Furthermore, after further investigation, he found out he would have
to purchase a "quicker" harddrive (he just bought a brand new PC
_just_ so he could get SAW). If you want more details, let me know.
I'll talk to my friend and find out exactly what wasn't right.
I wasn't too impressed with it when I downloaded the demo..
toehser@cais2.cais.com (Tom Oehser) writes:
>Also, what is the limitation that the demo holds that the full
>product doesn't?
Uh... guessing... older version... limited to 1 minute... no smpte
sync...
It is US$400 from SoundWare
jmireau@tibalt.supernet.ab.ca (James Mireau) writes:
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I was reading [about] SAW when I came across the story .. [of]
hardware woes.. [Some guy] had recently bought a new computer system
for specific use with SAW. Then he discovered that he could only
record two tracks and required memory and hard drive upgrades. What
was his system lacking? (I've also heard rumours that SAW requires
extra toys to function at full capacity.)
(**CAUTION** The information below may be totally erroneous. It is an
amalgam of the fruits of personal research, the advice of friends and
potentially bull-sh*tty rumours. Please e-mail me with corrections or
expressions of disgust. Proceed.)
I've picked the brains of a few friends and learned that:
1. You need an advanced IDE card for your computer. I guess
motherboards made earlier than last fall can't access hard drives
larger than 540Mb. An advanced IDE card will access up to 1.08Gig and
function quite quickly. A friend claims that advanced IDE has a data
transfer rate that is even faster than SCSI (though SCSI drives can be
much larger than IDE drives).
2. You should go with a decent soundcard. According to a
hardware/software guru-friend, all current Soundblaster cards are
incapable of simultaneous recording and playback (not to mention that
their S/N ratio kind of bites). I saw this claim contradicted in a
recent issue of Keyboard magazine (Nov 94?). I'm still trying to
figure this one out. Apparently, Turtle Beach's Tahiti card works
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nicely with SAW. I also hear that SAW has been optimized for a
semi-obscure professional sound card called CardD+ ($1100 Canadian).
3. As far as memory goes, I heard that 8 megs of RAM is sufficient.
Please let me know [people's experiences getting the right hardware
for SAW]. I am seriously considering buying SAW and I would like to
know what I'm getting myself into. It retails here for $700
(Canadian), and I'm on a student budget. If I need to sink another
$1000 into hardware, it's game over for my digital recording dreams.
Q1.7.3 What is a MIDI sync device?
A "sync" (synchronization) device does two things:
- When you play a sequenced MIDI set of backing tracks to the sync
device, it will generate an audio signal that specifies the timing of
the MIDI. This signal (which is not at all musical, it sounds like
white noise or a modem) can be recorded.
- When the recording is played back to the sync device, it will
generate timing MIDI data to drive the sequencer.
In short, it allows you to synchronise a MIDI track (or many) being
played by a software or hardware MIDI sequencer to a 4-track recorder.
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You require one track of the 4 for the sync signal. You don't need to
record the MIDI parts, since you can play them in sync with the tape
while you record the guitars, vox, etc (all the 'human' parts) on the
other three tracks of the 4-track. Finally, master the whole lot in
sync onto your DAT or whatever.
Remember, the *key* thing is that the MIDI parts don't ever need to go
onto any tape until you master. This means that you can adjust synth
balances, levels, etc right up until the final mix, and also means
that the synthesised parts will not suffer any degradation due to
being on tape; they're always 'first generation'.
Once again, the process is as follows:
- Record the MIDI sync track on your 4-track machine with a sync
device, having a MIDI device playing through the sync device
- Record the human tracks while using the sync device to play the
MIDI. The MIDI is not recorded; it is used by the humans performing.
- Play back human parts and MIDI parts (with sync device) onto a
master tape
Q1.7.3.1 What is an SMPTE-to-MIDI converter?
What is a JL-Cooper PPS-1??
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What is a JL-Cooper PPS-2??
What is a JL-Cooper PPS-100??
What is a Pocket Sync??
talarczyk@ACFcluster.NYU.EDU (Michael J. Talarczyk) writes:
These devices are SMPTE-to-MIDI convertors. While I don't know what
the pros in the studios would say about this, my personal thought is
that I wouldn't call them synchronizers, since there are machines that
sync one or more slave tape decks up to a master SMPTE source by
varying the speed of playback. (Although the JLCooper catalog I have
calls them synchronizers anyways ;)
I have a JLCooper PPS-100, which is a rack-mount version of the PPS-2.
Both devices take an audio input that is streaming SMPTE (Society of
Motion Picture and Televison Engineers [time code]) and converts that
to a MIDI output, which streams out MTC (Midi Time Code.)
Since I do film scoring, I'll tell you how you use this in a scoring
application.
If you have a master video (on a VHS or Beta or whatever tape,) you
stripe one channel of audio with SMPTE code. This gives every frame
of video a corresponding time in hours:minutes: seconds:frames
[e.g. 01:02:39:15] and is expressed by the audio coming from said
audio channel.
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This audio output is plugged into the PPS-100 (or PPS-2) and the
output of the PPS-100 is plugged into some MIDI port on your rig.
Assuming the software is set up right (which I won't tell you how to
do in this post!) you can follow the exact frame of video your VCR is
on, on your sequencer or keyboard.
So, when this is set up on my computer with Cakewalk, I can start the
video first, then start play on Cakewalk. It listens for the MTC code
coming from the PPS-100 (which is listening to the audio channel of
the videotape) and plays *exactly* at the spot in your sequence that
corresponds to the video's time frame.
Loosely, the difference between the PPS-2 and the PPS-100 is that the
PPS-100 is programmable, gives you two pulse relay contacts for
triggering non-MIDI devices (lights, klaxons, whatever...) and is
rack-mountable with an LCD panel and buttons instead of tiny levers.
michaelt@nylink.org (Michael J. Talarczyk) writes in a 2nd message:
Let's say you have this rig all set up, and you want to do a demo of
your sound design prowess with your new equipment. Now, let's say I
tape a commercial of, oh, a Scope commercial from regular TV. My VTR
(video tape recorder -- when you buy pro gear you have to be a snob
and call it something to distinguish it from being a VCR ;) has the
ability to selectively erase the sound on a videotape while leaving
the video alone. Let's say I remove all trace of sound on the 15
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second clip, and leave just pure visual.
Let's now say that I take my fab VTR and stripe one channel of my
little videotape. I now have a videotape with a lot of funny sounding
noise -- BUT, I can use that noise with my rig to put my own sound in.
Ok, now I have something like this:
VTR --audio--> PPS-100 -- MIDI (MTC) --> Computer
Since the video tape is playing SMPTE time code, and that time code is
being interpreted by the computer via Midi Time Code, my computer
knows what frame of VIDEO I'm on, every time I record something with a
MTC-aware sequencer.
Cakewalk is one such sequencer. It allows you to compose and edit
music in a number of graphical formats, and allows you precise editing
and control of your MIDI sounds and equipment. Assuming my
connections are right up 'till this point, when I press the Record
button on my sequencer, it knows what frame of video the VTR is
playing. This by itself is not really neat or anything, but the
advantage becomes clear when you PLAY the sound.
Locking the sequencer to the video tape means that if Cakewalk has a
cue for a cymbal crash at frame 01:00:01:15.13
(reel:hour:minute:second:frame), it plays a cymbal crash when the VTR
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hits that frame.
So, going back to our Scope video. Let's say that at frame
01:00:01:15.13 we have a huge wave crashing over the screen. We think
we have a nice cymbal that approximates that sound (ok, samplers are a
different post.) If we rewind the video tape to, hmm, maybe 10 or 20
seconds before the wave visual, press play on the video tape, and
press record on our sequencer, we're recording in sync. Now, when the
wave crashes, you go BANG on the keyboard (or whatever gear triggers
your cymbal,) and you hear a nice crash, at the exact point where you
see the wave crash over the screen. You wait a few seconds for the
sound to die away, and you stop all the machines.
Now, if you play back the video tape and SLAVE the sequencer to it,
the sounds on the sequencer will play back in EXACTLY the same place
as you recorded. Thus, your cymbal will crash every time you see that
wave crash over the screen, with approximately 1/30 sec accuracy. You
can build up multiple sounds and effects to recreate the entire sound
of the original videotape, using whatever sounds you want. With a
sampler (or an analog deck, for pre-Gen-X'ers,) you can even add your
own dialog this way.
Q1.7.4 What is Sounds Effects?
A public-domain Macintosh multi-tracking program. See Q1.7.
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[How is it?? Where do you get it?? --DSF]
Q1.7.5 What is DigiTrax?
A low-cost Macintosh multi-tracking program. See Q1.7.
[How is it?? Where do you get it?? --DSF]
Q1.7.6 What is Cakewalk?
Cakewalk is an MTC-aware sequencing program. See Q1.7.3.1.
[How is it?? Where do you get it?? --DSF]
ram@indigo3.carb.nist.gov writes:
There are two versions of Cakewalk. One called Professional and one
called Home Studio (I think). The Professional one is more expensive,
and better. It lets you play .wav files at the same time as playing a
MIDI file, which is quite useful for doing PC recording where you can
play.
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Q1.8 How does an 8-track machine compare to a 4-track machine?
Clearly, they have 8 tracks instead of 4. However, they are also
aimed at a more high-end market. For example, I was unable to find a
4-track machine which used DAT (Digital Audio Tape). I found,
however, the Yamaha D88 8-track, which lists at US$4499.
The Tascam 688 Midistudio 8-track lists at US$3779.
tstrohma@theodolite.ae.calpoly.edu (Trevor Strohman) writes:
The ADAT is now available from Alesis and has been proven in the pro
marketplace. Prices are hovering above $2000. This type of machine
needs an external mixer. Multiple units can be synced together. Many
studios use these, so if you happened to buy one you would be format
compatible with the industry. However, if you're going to spend that
much, you'll need a mixer and good mics... Soon you're out $5000.
Q1.9 How should I clean my 4-track machine?
dragon@homerecording.com (Dragon) writes:
>llinimon@aol.com (L Linimon) writes:
>1) Is it better to clean with a kit (aka swabs and head cleaner,
>etc.) or one of the tapes that head cleaner is placed on? The cleaner
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tape is "played/recorded" on, and the liquid makes slight contact
>with the "dirty" bits.
Use the kit, where you can see what you're doing. The tapes are better
than nothing.
>2) Is the rubber washer (or whatever it is) near the capstan a high
>maintenance item?
It's the pinch roller. Get all the oxide off it and use rubber cleaner
on it.
>3) What is the best way to demag...with a Radio Shack Demag tape or an
>actual heavy-duty degausser?
I'll leave this for the flamefest that will no doubt erupt again...
ak748@detroit.freenet.org (Daniel W. Newport) writes:
Use *only* rubber cleaner on pinch roller. Use a kit to clean with
lintless foam swabs.
Q1.9.1 Do I need to demagnetize my 4-track heads?
[The newsgroup readers appear undecided. Read on. --DSF]
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psu02178@odin.cc.pdx.edu (Joshua Meredith) writes:
Demagnetizing on a *cassette* four track is not necessary. Due to the
slim width of the tape and the relatively slow speed of the transport
(even when you're using double speed), magnetism does not really
accumulate. On top of that, every time you put the machine in record,
it demagnetizes the heads so even if you are concerned about it, the
machine does it for you.
With reel to reel tape decks (stereo or multi-tracks), demagnetization
is very important, as the accumulation is too much to be eliminated by
simply engaging record.
kowalski@scooter.ping.de (Thorsten Kowalski) writes:
I use a degausser myself, did cost me about US$20 but there are some
that cost US$60. I don't know if they are better. Mine's very useful
for any kind of tape machine. I prefer it over a battery-powered tape
unit; those are only good if you can't use a degausser, e.g. in a car
radio.
Degaussing should be performed about every twenty hours of usage.
I use a degausser with my home cassette deck, a Nakamichi. There were
some people who asked me if they should go and buy such a device and I
gave them mine so they could test. They have never done so in about 2
years and after degaussing they have lots of treble which they weren't
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missing because the loss of treble comes slowly.
I've never heard of any tape device that degausses itself. If there
are some please give me their names.
jeibisch@revolver.demon.co.uk (James Eibisch) writes:
So why does the manual for my Tascam 05HS (cassette), and everyone
I've heard comment on the subject recommend degmagnetising frequently?
S2. Effects
Q2.1 What is flange?
benl@mojo.europe.dg.com (Ben Last) writes:
Roughly speaking, what you'd get if you recorded the same guitar onto
two tape recorders at once and during replay, slowed one of them down
*just a little* by putting your finger on the reel. A bit like a
chorus.
This is *not* a general effect with which you'd treat a whole mix.
[Is this explanation okay?? I thought it produced a somewhat
time-varying signal that sounded a bit like a slow "wa-wa"
sound. --DSF]
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Q2.2 What is reverb?
benl@mojo.europe.dg.com (Ben Last) writes:
The technical explanation would be something like: "a composite of
echoes from many sources, including early reflections from nearby
surfaces and remote echoes of longer duration from distant surfaces."
Ben's Patent Simple Explanation is: think about standing on the stage
of an empty theatre (theater for Americans :-). Shout. Imagine how
your voice would sound; sort of echoed back from lots of different
places. Consider how your voice would sound if you were inside a
wardrobe (closet). No echoes at all. The one with the echoes has
more reverb.
Q2.3 What is compression?
Adapted from stabnste@phoenix.phoenix.net (Erik Karlson):
A compressor reduces by a preset ratio the level of any incoming
signal which exceeds its preset level. For example, if your
compressor is set for 10 db and has a compression ratio of 2 to 1, any
signal that comes in over 10 db would be reduced by a factor of 2. For
every 2 db of input signal over 10 db, the output level would only be
increased by 1 db. This is very useful when recording vocals because
they tend to have varying levels. As with all things, too much of a
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good thing can be bad. Too much compression can leave your sound dull
and choppy.
pseo@mail2.sas.upenn.edu (Peter S Seo) writes:
[S]ay you're playing the guitar for instance, and for some reason, the
dynamics (loudness, volume, whatever) of each pluck is different maybe
because you're just really bad at playing guitar. (btw, i'm not
inferring that [only bad guitar players] use compression...)
[Compression will] "smooth" out the signals, the plucking, so that
each pluck will be of the same output level or volume.. it makes the
signal much "tighter" i suppose.. this helps out alot for vocals
too.. but it is something that you the musician have to decide to
use.. maybe you don't want it smooth..
you can also use a compressor to sustain a guitar signal.. though someone
else will have to explain how that works.. all i know is that it does.
benl@mojo.europe.dg.com (Ben Last) writes:
Same way as it sustains everything else: You pick the string. The
note starts, and as it goes on, it gets quieter. Without a
compressor, it'd die away. What the compressor does it to keep
'turning up the volume' so that the note appears to stay the same
level. Eventually, the volume (amplitude) of the note coming from the
guitar gets so quiet that the compressor gives up and turns the volume
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ack down.
Similarly, a compressor can reduce the volume when it hears notes
which are too loud; the end result is that the volume level coming
from the compressor is much more constant than without it.
Q2.3.1 Do I need stereo compression?
> jthan@world.std.com (Jonathan M Richardson) says:
> My local music shop recommended a dual channel rack mount unit which
> costs $180! Is there a cheaper solution?
adouglas@belvoir.com (Andrew Douglas) writes:
I have little experience with recording, but the conventional wisdom
is that you really do need a stereo compressor. You compress vocals
and bass, and in your case acoustic guitar, while laying the track
down, then perhaps add just a touch during mixdown. (which is the
reason for stereo)
A stereo unit will also allow you to put different amounts of
compression on the two channels.
brianb@scorpion.iii.net (brian q. buda) writes:
Well, for one, who makes the compressor? $180 is really cheap. [Y]ou
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can save some money by going with a mono compressor. Stereo is all
well and good but for small 4-tracks it has no use. That is if you
don't mind compressing down to track and not on mix down. Stereo is
really only for live gigs and when you have a multi-track system with
several AUX sends so that you can patch each channel into a mix.
Q2.3.2 Should I use compression on drums? Vocals? Bass?
ertrinid@girtab.usc.edu (Elson R. Trinidad) writes:
If you mic up your kit and route it through a separate mixer before
going to your 4-track, put the snare through a compressor (the kick,
too, but through a separate compressor) and you'll have a more
"professional" sound. But never try to compress the entire drum mix,
especially if you use cymbals a lot - there will be a lot of
unnatural-sounding "pumping and breathing".
brianb@scorpion.iii.net (Marc?) writes:
It's great for snares and kicks. (fattens up the kick alot) But I find
that it kills rides and hihats. To really compress drums you need to
have a system that can compress some, but not all.
Also, [noise] gates are needed to really make it sound right.
You need different amounts of compresson on different things. That
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is, vocals don't need as much as a bass does.
Q2.4 What is limiting? How is it different from compression?
ak748@detroit.freenet.org (Daniel W. Newport) writes:
It's compression in the extreme. That is, once the signal [level]
reaches some preset threshold instead of compressing it at some ratio
say 2:1 or 4:1, [the limiter] limits it at that level, (infinity:1) by
not allowing the signal level to go beyond this.
lwillia@ix.netcom.com (Larry Williams) writes:
Some limiters will allow a slight increase in level above the
threshold (a ratio of 20:1, for instance), others act as a "brick
wall" limiter allowing absolutely no level above the threshold.
Limiters are useful especially on digital audio equipment where no
signal above 0 VU should ever exist. They are also useful in live
sound systems where feedback volume can be controlled, protecting both
the speaker systems, and the performers' and listeners' ears.
[What is a Dietz dual limiter?? --DSF]
Q2.7 What is a noise gate?
A noise gate lets a signal through when it is louder than a certain
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level and cuts it off when it drops below that level.
Imagine you play a chord on your guitar, which is connected to a noise
gate, which is then connected to your amp. The noise gate 'sees' a
decent signal level and lets the sound through. You then damp the
chord off with your hand.. The noise gate sees the signal level drop
and cuts the sound off. If the gate weren't there, you'd still hear
all the low level stuff like hum, the sound of your hand sliding on
the strings, the sound of the guitar clanking against your belt
buckle. As soon as you play a note, the gate 'opens' and you hear the
guitar again.
Q2.5 What is (upward) expansion?
marcl508@hudson.iii.net (Marc LaFleur) writes:
Upward expansion is the inverse of compression. Where compression
decreases excessive peaks over a threshold, upward expansion decreases
excessive troughs below a threshold. As the signal falls below a
preset lower threshold, the gain is scaled "upward" by the set ratio
(e.g. an input signal falling below the threshold by 10dB would be
boosted to 5dB below the threshold with a ratio of 2:1).
Q2.6 What is companding?
Adapted from lwillia@ix.netcom.com (Larry Williams):
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Companding is a two-stage process of COMPression and exPANsion which
reduces the lowest noise levels on the tape. Companding typically
refers to a tape recording process (instead of live sound or effects
processing). For example, a signal may be compressed at a 2:1 ratio
when recorded, and expanded at a 1:2 ratio when being played back.
This is also known as double-ended noise reduction, meaning that one
has to have a decoder to listen to the encoded signal, contrasted with
single-ended noise reduction where no decoder is required.
Q2.8 What are some brands of effects boxes?
brand effects price
--------------------------------------------------
Boss SE-50 ?? ??
Lexicon LXP-5 ?? ??
ART FXR Elite ?? ??
Alesis Midiverb III ?? ??
Digitech TSR-12 ?? ??
DSP 256XL ?? ??
See also "Home & Studio Recording," March 1994, page 45.
Q2.9 Should I get an all-in-one effects box or a more basic box and
a separate compressor/limiter?
dragon@homerecording.com (Dragon) writes:
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[S]hould I try to get a super-duper processor that has all this
>stuff in there at once, or should I get a more basic efx box and a
>separate compressor/limiter? The latter course seems more logical,
>but some of these units (like the Boss SE-50, available used for
>decent prices) seem to have an awful lot of stuff in them. OTOH, Boss
>won't quote noise figures for their units, which makes me quite
>suspicious...
>> OK, here I am following up my own article, but I've already
>>found a good article on this very subject in Home & Studio
>>Recording, March 1994, page 45, which says that you should go
>>for the separate compressor/limiter. That's what I
>>suspected. So I guess I'm also looking for recommendations in
>>that area.
taroh@kohnolab.dnj.ynu.ac.jp (Dr. Taroh SASAKI) writes:
First of all, a good flanger (also distortion) inevitably
makes a lot of noise. I tried some multi-effectors of low price, and
[recommend] BOSS SE for your purpose. SE has analog [circuitry] and
its effect is very good for guitarists. Also [has a] low price. I
know (and own) a SONY MP, which processes [effects digitally]. Maybe
pitch shifting, comp/limiter and so many ``clean'' effects work better
than SE, but franger [flanger?], distortion, ... ``dirty'' effects
[are] not vigorous. MP's much better in multi-tracking; maybe SE's
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sound becomes [dirty] when you repeat ping-pong-pung-peng-... [^_^],
but in guitar playing.
cheyenne@netcom.com ("Will" using the account of Melissa Duncan) writes:
What works for me is my Digitech RP-1 effects processor. It doesn't
have the pitch shifter but it has the other stuff you wanted, and I
think you can get them new for around US$400. I bought mine for guitar
originally but have had good success running vocals and bass through
it, and even effecting my drum machine with it. I don't think it's
noisy at all, it has been an infinite improvement over my Ibanez stomp
box collection.
dragon@homerecording.com (David Fiedler) writes:
Actually, I went out and got a Digitech TS12. And until it crapped out
on me after 8 hours or so (just before I started recording with it, of
course), it sounded great. But I must warn anyone thinking of getting
one that it is NOT the "little brother" of the TS24 as it is hyped to
be...it does not have anywhere as much sampling memory, and does NOT
do separate processing on both sides like the ART FXR can.
Funny thing, about a year ago (or was it 2?) I bought an RP-1 and sold
it on the net a few months later, as it was overkill for the guitar
stuff I was doing. I was remembering it the other day...it was *lots*
noisier than the TS12...I guess that new circuitry or software or
whatever really does work to cut down the noise. But I think a lot of
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that RP1 noise was frm the compressor and analog section, not the
digital. Anyway, you're right, it's certainly better than stomp boxes,
although a lot harder to tweak in real time.
benl@mojo.europe.dg.com (Ben Last) writes:
You need to consider the difference between 'inline' (or 'insert')
effects and 'in loop' effects. The difference is essentially 'how
much of the original signal is passed through the effect?'. For
example, with compressors, limiters, eq's, etc, what you get out is
essentially the original ('dry') signal modified. With reverbs,
chorus, flange, distortion, you can usually set the effect to produce
only the modified ('wet') signal.
Units that just modify the signal need to go 'in line' between the
instrument and the mixer / deck. You can't use them in an effect
loop. Units that can return just the treated (wet) signal can be put
either in line or in a effect loop.
Thus, if your setup includes an effect loop and you get an all-in-one
box including compression, you won't be able to use the compression
whilst the unit is in the effect loop.
S3. Recording Techniques
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Q3.1 How do I get a good sound?
David Copeland (davidc@access.rrinc.com.blacksburg.va.us) writes:
- Try to record intruments seperately (unless you have lots of rooms
to isolate the instruments)
- Mike the bass. Plugging a bass straight into a four track usually
leaves a dull thud bass sound. If you don't want to mike it, try to
use an eq pedal or something to shape the sound and boost the signal a
little (or if you have active pickups...)
- To get a really good guitar sound, this is what I try to do.
o Record the drums and bass onto two tracks.
o Have the guitar player play the guitar track on both of the
remaining tracks separately (so there's two takes of the same track)
o Then, put the bass and drums center and one guitar right and one
left. It creates a really cool chorussy sound and makes the guitar
extra heavy.
o Then mix down onto another cassette. The new cassette has
everything but the vocals on two tracks leaving you with two more
tracks to do vocals, leads or whatever. This process can be repeated
for more tracks.
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Frank Zappa recorded an album on a 4-track this way and got 48 tracks on it.
- Try to record the guitar at "live" levels. Most amps really shine
when cranked up.
- When doing the final mix, if you have a stereo chorus, you can give
the tape a more stereo sound. Also reverbs and delays on snare,
vocals, guitar (depending on guitarists rig) are good.
mcknight@pire.org (Scott McKnight) writes:
Another approach, and one that's the opposite of that mentioned by
Dave Copland in another post, is to try to get a good live recording
of as much of the band as possible in order to avoid generation loss
due to bouncing.
Basically what you do is to put a couple mikes in front of the drums
in an X-Y configuration. One mic goes to one channel the other to the
other. Find a way to get the bass in the center of the mix, either by
putting the bass amp behind the mics, across from the drummer, or by
running the bass direct and record onto the same two tracks that the
mics are recording on with the bass panned center. Put the rhythm
guitarist off to one side. Experiment with distances from the mics
and volumes until the stereo mix of these three instruments is good.
Record.
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Add lead guitar to one of the remaining channels and vocals to the
other. You could add lead guitar as part of the original 2 band
tracks by putting his/her amp on the other side of the mix (across
from the rhythm guitarist) but this eliminates the ability to punch.
Possible changes/additions to this system include adding solos by
punching on to the vocal track when there are no vocals, adding other
percussion (e.g., tambourine) by having friends standing in the room
playing them at appropriate distances from the mic while recording
other tracks, adding instruments by mixing them in while mixing down.
This method of recording your band gives a fairly natural, hifi sound
by eliminating generations and phase cancellation created by having
too many mics open in the room at the same time. If your not sure
what an X-Y config. is ask someone because doing this part correctly
is important.
mparrott@kendaco.telebyte.com writes:
If you have the extra bones, get a Tech 21 SansAmp GT2 (street price
about $175). It's a little stomp box that emulates no less than 27
different tube amplifier/amp modification/cabinet-mic placement
combinations, from nice 'n clean to balls-to-the-wall crunch... not
including the variations possible with the post-EQ, Gain, and Level
controls. It does a great job of simulating the sound of various mic'd
tube amps and cabinets in a 'dead' room. It's very expressive and good
not only for direct input of guitar, bass, keyboard, and vocals but
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also for livening up tracks when mixing down. The best thing about it
is you can get the sound of a full-bore Marshall stack when you want
it without disturbing everyone in the neighborhood. :)
When mixing down, let your fingers ride the volume controls for each
track, making EXTREMELY small adjustments in the volume level of each
throughout the song. It's hard to explain why, but it really livens up
a mix and keeps it from sounding flat. Less is more in this case; if
the adjustments are too extreme (to the point you consciously notice
them) then the mix just sounds screwed up.
Q3.2 What should I know about speeding up recordings?
schmange@wbb.com writes:
One secret George Martin always used to use was to speed up the tape
slightly for the finished masters. (All the early Beatles stuff is
speeded up slightly to make them sound more up-tempo)
bwill@teleport.com (Brad S Williams) writes:
Do recording studios have equipment to change pitch *without* changing
speed (so that I can speed up the mix while lowering the pitch to
offset for the chipmunks effect)? Do people do this kind of thing?
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will@iglou.iglou.com (William M. Willis) writes:
How much of a speed increase are you going for? I use the varispeed
or pitch control knob (whatever you want to call it) on my 4-track to
slightly increase the speed on most of my recordings. Actually though,
I didn't start doing it to make the songs more uptempo, but rather to
compensate for tapes stretching out and slowing down after numerous
plays. But anyway, I don't get any sort of munchkinization of my
vocals or instruments when I do this...but I'm also not raising the
speed more than a couple of cents.
benl@mojo.europe.dg.com (Ben Last) writes:
The keyword here is 'slightly'; more than a few percent and it's
chipmunk time. It's used reasonably frequently (IMHO) 'cos by the
time the final mix is made, those making it have heard the song
numerous times and it's getting a little stale. Speeding it up a
little gives it a bit of a 'push'.
dragon@homerecording.com (Dragon) writes:
It's trivial with digital processing software.
Q3.3 What should I know about using a VCR as a digital recorder?
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schmange@wbb.com writes:
What I've been doing lately .. is mixing 4 tracks down to a Pulse Code
Modulator (Sony PCM701es) and my VCR. It makes your VCR (even a mono
one) work like a digital recorder by converting your audio signal to
digital. You can mix 4 tracks down to the PCM/VCR and then re-record
it back to the 4 track in stereo (channels 1 & 4 panned far left &
right). There's minimal generation loss cause it's digital. Then you
can add more stuff on channels 2 & 3 and repeat the process or just
use the PCM as your digital master when you mix down.
Michael Crowl writes:
>On 4 Mar 1995, Giles. Harney wrote:
> Could somebody please tell me why would people want to
> mixdown/master/record to vcr tape? What are the advantages of
> doing such a thing, and how?
The audio portion on Hi-Fi STEREO vcr tape has a higher bandwidth than
conventional cassette tape, so the sound is much crisper and clean.
However, I have not tried this myself, as I do not have a Stereo vcr
deck. A friend of mine uses one to mix down and he's quite happy with
it and wouldn't even get rid of it for a DAT! Although, maybe he'd
use them side-by-side. He played a few tracks that he had done, and
then he played an XTC album that he had recorded straight off of vinyl
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onto his Hi-Fi deck. I have to say, it sounded lovely - all the
percussion was coming through perfectly thick (which digital can strip
sometimes), and it even seemed to limit some of the vinyl crackles
that came through.
jyk7853@is2.nyu.edu (Jason Y. Kaneshiro) writes:
One thing I've noticed which you may want to look out for - is that
some HIFI VCRS (if not all) have built in compression which may or
may not be desirable. As it's in the VCR you can't bypass it which
pretty much makes it useless if you want to avoid that kind of thing.
I still think DAT is the way to go...
Also, if you are making duplicates at a plant I think they would be a
little confused if you offered them a VCR tape and told them to press
your CD or make your cassette copies off of it... the majority of the
stuff is on DAT, and I think even one of those $400 portable walkman
sized DAT things would sound better than a VCR or cassette master.
reedijk@gene04.med.utoronto.ca (Rob Reedijk) writes:
[S]ome of your contributors mentioned Hi Fi VCR as a good format to
mix and bounce to. This is true in that Hi Fi VCRs do have S/N ratios
in the 90 db range (comparable to cd). Where you have to be careful
is that since you can get Hi fi VCRs for $200-300 some of them will
certainly have shit pre-amps. Take it case by case. You probably pay
for what you get.
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Q3.4 What should I know about doing an external pre-mix versus
an internal bounce? (Or "Using a second recording device for
mixdown?")
An "internal bounce" is using the 4-track itself to reduce the number
of tracks.
An "external pre-mix" is playing to an external device (e.g. VCR, DAT
recorder) in stereo, then playing that recording back into the 4-track
onto two tracks. This leaves two tracks free.
sratte@mindvox.phantom.com (Swamp Ratte) writes:
Somebody was asking why you'd wanna do an external pre-mix and return
it instead of doing an internal bounce. Well, I've recently started
running into problems doing bounces with my Tascam Porta Two.
This band's demo I've been doing, we definitely want a huge gigantic
wall-of-fuzz-guitar sound. Thing is, it's just one guitarist with no
real leads, just clean parts and dirty parts (it's pop-punk music).
So we laid a track of all-clean, straight through the songs. Then set
up two half-stacks with different distortion sounds and EQ setups on
each one. Ran a line splitter on the guitar out so one went to both
amps and one line went into a SansAmp and then was mixed in with the
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miked amps and put on another track. So that's 3 guitar sounds on the
one track, plus the clean...and when I bounced to one track to make
room for bass and vocals, I added SansAmp dist. to the clean track
also...5 guitar sounds total on dirty part. While testing out the
bounce, it was HUGE...especially panning different parts all over the
place, rad as can be.
The actual bounced track though? Ugh. At certain parts the guitars
seem to fade in an out, along with the cymbals...horrible phase
problems, too much stuff at certain freqs, etc. i guess.
However, if i set up a nice stereo mix and record it to my mixdown
deck, then run the lines back in to two tracks, I get the huge stereo
thing with no problems, and two tracks left over.
Plus, it's nondestructive... it's a drag if you do a bounce and then
later decide it sucks and have to tell the band, "Well, you're gonna
have to bring the drummer back with his drums and do it all over
again, 'cause this bounce mix is bad. Uh, sorry." This way, you've
still got the original 4 tracks to mess with again if you want.
me@ram.org (Ram Samudrala) writes:
[Think about using] a 2nd deck or DAT recorder (preferred) for
mixdown. I believe I say why this is good [in my recording tips Web
page-- see below] in terms of elimination of noise, but it is also
good to keep the original tracks around so you can remix and re-edit.
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You'd be surprised at how well this technique works.
Q3.5 Can I use compression on a mixed signal?
Adapted from what robertb@primenet.com (Robert Blackwell) writes:
>billrich@caspian.ext.vt.edu (Bill Richardson) says:
>Can you use a compressor on a mixed signal, like between the 4 track
>and the mastering deck?
Yes you can. In fact that is what you want to do to keep [everything]
in its perspective position. [As] I have said before, it should be
used very carefully, if you [over-compress], the mix will [lose] it's
"live" feel and it won't have any dynamics.. It is also good at
keeping the signal going out to the mastering deck from getting too
hot and overloading the circuitry ([with noise]).
Q3.5.1 Is it better to compress when recording or mixing?
mcknight@pire.org (Scott McKnight) writes:
My stock answer is "depends on how much compression you have." If you
have enough channels of compression to compress everything at mixdown
then you might want to hold off compressing til then. Advantages of
doing this include being able to compress based upon hearing
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everything at once and not making any compression decisions that you
can't undo. Disadvantages of this include losing a certain amount
noise supression gained by being able to record the compressed signal
hotter overall. If you don't have enough compression to process
everything at mixdown that you want to compress, you'll have to
compress some things when recording.
Q3.6 How do I improve my drum sound?
mcknight@pire.org (Scott McKnight) writes:
I tend to rely on the overheads for the basic sound of the drums and
use the individual drum mics to control the mix a little better and to
be able to put effects on individual drums.
You might also try improving the drum sound by cutting out a
generation, i.e., instead of recording to multiple tracks and mixing
down to 2, try mixing straight to 2 or 3 tracks while recording. I
like to record 3: drums left, drums right and snare. It takes a
little more time to experiment getting the mix right (especially if
you have no control room or engineer) but IMHO it's worth it.
Most articles I've read recommend using as few mics as you can get
away with. This means that everything may not get close mic'd but
I've always preferred the sound of drums mic'd from a few feet away
anyway.
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Q3.6.1 How would I use noise gates to improve my drum sound?
mcknight@pire.org (Scott McKnight) writes:
If you are mic-ing a lot of drums individually I'd try using noise
gates on the toms, snare and maybe kick. This is to cut down on phase
cancellation due to having all mics open all the time. Try listening
to your drum tracks at some point where it's a basic beat, no toms.
Listen to the cymbals while bringing the tom mics in and out of the
mix. When I do this I find the cymbals (and for that matter the whole
kit) sound better when the tom mics are out of the mix. By gating
each tom individually you make so that each mic is only open when you
need it.
S4. Related information
S4.1. Information online
Q4.1.1 What are some related newsgroups?
comp.dsp - Digital Signal Processing
rec.music.makers.songwriting - Songwriting discussions
alt.music.lyrics - Lyrics to songs by request
more??
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Q4.1.2 What are some related FAQ lists?
Blurbs given below:
Archive-name: AudioFAQ/pro-audio-faq
Archive-name: dsp-faq
Archive-name: music/midi/bibliography
Archive-name: music/midi/archives
Archive-name: music/composition-FAQ
Archive-name: music/netjam-faq
DAT-heads microphone FAQ and other DAT-heads FAQs
Archive-name: law/Copyright-FAQ/part1
[I know there are more. Help here?? --DSF]
****************************************
Archive-name: AudioFAQ/pro-audio-faq
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:
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ec.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.
Here's a hypertext reference to an HTML file:
rec.audio.pro
FAQ (v2.14)
****************************************
Archive-name: dsp-faq
You can ftp the very latest version of this FAQ from: evans.ee.adfa.oz.au
(131.236.30.24) in pub/dsp/dsp-faq.help, or from copernicus.Berkeley.EDU
in dsp-faq.help.*.
Q0: What is comp.dsp?
Comp.dsp is a worldwide UseNet news group that is used to discuss
various aspects of digital signal processing..
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****************************************
Archive-name: music/midi/bibliography
------------------------------------------------------------------------------
Version: $Id: bibliography,v 1.25 1994/09/08 12:24:00 piet Exp $
------------------------------------------------------------------------------
This is a bibliography on sy [OOPS .. accidentally clipped and lost]
.. structure into it, but not all books will fit into a single
subject. NOTE: I haven't read these books, and the comments are from
other people. On some of them I lost the original commentor's
name. Sorry about that. If you have additions or correction to t
[OOPS].. There is a more scientifically-oriented bibliography
available in the Computer Music Journal archives, on the ftp sites
mitpress.mit.edu /pub/Computer-Music-Journal/EdNotes or
ccrma-ftp.stanford.edu /pub/Publications/cmj/EdNotes. Some references
in this file that belong in that category will be removed in the
future.
A bibliography on alternate tunings can be found on ella.mills.edu in
/ccm/tuning/papers/bibliography.
****************************************
Archive-name: music/midi/archives
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---------------------------------------------
Version: $Id: archives,v 1.121 1995/02/24 14:44:20 piet Exp $
Note: the latest version of this file is available from the ftp.cs.ruu.nl
archive as MIDI/DOC/archives (see below how to access the archive) and
ftp.funet.fi /pub/sounds/midi
ftp.ibp.fr /pub/midi
------------------------------------------------------------------------
****************************************
Archive-name: music/composition-FAQ
Last-modified: 1994/9/30
Version: 2.2
This is a music composition frequently-asked questions (FAQ)
document, distributed by NetJam. It is probably of interest to readers
of the USENET newsgroup rec.music.compose, and anyone else interested
in music composition.
It is posted fortnightly to the above-mentioned group, as well
as to news.answers and rec.answers. It is also available as
ftp://XCF.Berkeley.EDU/misc/netjam/doc/FAQ/composition/compositionFAQ.
Finally, it can be obtained by emailing NetJam-request@XCF with the
subject line "request for composition FAQ". The machine
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XCF.Berkeley.EDU has IP address 128.32.138.1.
Please send contributions and comments to
NetJam@XCF.Berkeley.EDU.
For general info about NetJam, email
NetJam-request@XCF.Berkeley.EDU, with the phrase "request for info" in
the subject line.
****************************************
Archive-name: music/netjam-faq
Last-modified: 1994/10/17
Version: 5.5
This is an electronic and computer music frequently-asked
questions (FAQ) document, distributed by NetJam. It is probably of
interest to readers of the USENET newsgrooups:
ucb.becmug
rec.music.makers
rec.music.makers.synth
rec.music.synth
rec.music.compose
comp.music
and anyone else interested in the applications of computers to music
(and vice-versa).
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It is posted fortnightly to the above-mentioned groups, as
well as news.answers. It is also available as
ftp://XCF.Berkeley.EDU/pub/misc/netjam/doc/FAQ/ECMFAQ. Finally, it can be
obtained by emailing NetJam-request@XCF with the subject line "request
for ECM FAQ". The machine XCF.Berkeley.EDU has IP address
128.32.138.1.
****************************************
http://www.atd.ucar.edu:80/rdp/dat-heads/
"This is the WWW home page for the DAT-heads mailing list. DAT-heads
concerns itself with Digital Audio Tape decks, with an emphasis on
their use for the recording and distribution of live music, as done by
the followers of the Grateful Dead, Phish, and others."
There are several DAT-head FAQs and related materials and Jeff Maggard's
http://www.ultranet.com/~jgm/dat-home.html
The blurb from the DAT-heads microphone FAQ:
This is a collection of information of potential interest to users of
microphones, especially those who use them to record live musical
performances. The information was contributed by the users of the
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DAT-Heads mailing list.
****************************************
FREQUENTLY ASKED QUESTIONS ABOUT COPYRIGHT (V. 1.1.3)
Part 1 - Introduction.
Copyright 1994 Terry Carroll
(c) 1994 Terry Carroll
Last update: January 6, 1994.
This article is the first in a series of six articles that contains
frequently asked questions (FAQ) with answers relating to copyright law,
particularly that of the United States. It is posted to the Usenet
misc.legal, misc.legal.computing, misc.int-property, comp.patents,
misc.answers, comp.answers, and news.answers newsgroups monthly, on or
near the 17th of each month.
This FAQ is available for anonymous FTP from rtfm.mit.edu [18.70.0.209],
in directory /pub/usenet/news.answers/law/Copyright-FAQ, files part1 -
part6. If you do not have direct access by FTP, you can obtain a copy
via email: send a message to mail-server@rtfm.mit.edu with the following
lines in it:
send usenet/news.answers/law/Copyright-FAQ/part1
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send usenet/news.answers/law/Copyright-FAQ/part2
send usenet/news.answers/law/Copyright-FAQ/part3
send usenet/news.answers/law/Copyright-FAQ/part4
send usenet/news.answers/law/Copyright-FAQ/part5
send usenet/news.answers/law/Copyright-FAQ/part6
quit
****************************************
Q4.1.3 What are some related email lists?
Blurbs below:
Mixmasters
GAJOOB Magazine's DiY Report
DAT-heads
**********************************************************************
Mixmasters is a worldwide group of people who exchange audio tapes and
advice for the sole purpose of helping improve each others' production
values.
For more information, send the command
info
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in the body of an email message (*not* in the Subject: line) to
mixmasters-request@netmaniac.com. To join the list, send the command
subscribe
in the body of an email message (*not* in the Subject: line) to
mixmasters-request@netmaniac.com. You can add an email address at the
end of the above line to specify where you want mail from the list to
go. And to send messages to the list itself once you're a member,
send them to:
mixmasters@netmaniac.com
Thanks for your interest, and have fun!
How it works:
You write to the list telling everyone about what kind of tape you've
made (in terms of music genre, or whatever the subject of the tape
is), and then privately correspond with whoever shows an interest in
receiving it. The people who get the tape listen to it, then post
their comments to the entire list. It's kind of like a do-it-yourself
version of Michael Laskow's "Reader's Tapes" column in Recording
magazine, except that none of us are Michael Laskow (who knows, maybe
we can get him in for a cameo one day :-).
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We also help each other with ideas on equipment and how to use it.
Join us! No cost or obligation...
--
David Fiedler Internet:dragon@homerecording.com Phone:530/677-5870
USMail: HomeRecording.com, PO Box 220, Rescue, CA 95672
**********************************************************************
gajoob@utw.com (by way of gajoob@utw.com (Bryan F. Baker)) writes:
HOW TO RECEIVE THE DiY REPORT
GAJOOB Magazine's DiY Report is available in a free electronic
edition, e-mailed 2-3 times each month, and available by sending
e-mail to: GAJOOB@UTW.COM. There is no automated list server set-up,
so just send a regular e-mail message telling me you want to subscribe
to the list and you'll start receiving with the very next mailing. You
may request a free sample copy of the paper edition of the DiY Report,
which is published in conjunction with the electronic edition
(information is identical in both editions).
BACK ISSUE ARCHIVES
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Currently, archives for GAJOOB Magazine's DiY Report can be found on
the World Wide Web (complete with hypertext links), courtesy of DiY
artist Philip Riley, at: http://www.math.duke.edu/~priley/
GAJOOB MAGAZINE's DiY REPORT INTRODUCTION
Established in September 1994, GAJOOB Magazine's DiY Report is
published 2-3 times each month and contains news of outlets for DiY
artists, such as compilation projects in the works, collaborations,
radio shows, publications, distributors and others. The DiY Report
also contains listings and reviews of the latest DiY releases
(including your own!), whether it's a homemade cassette offered to the
general public or a larger run of CD's. There are also discussions of
relevant topics, including a letters section, articles offering tech
advice and interviews with DiY artists talking about their
craft. There is also some coverage of indie bands as the distinction
between purely DiY and "indie" becomes more and more blurred.
**********************************************************************
DAT-heads concerns itself with Digital Audio Tape decks, with an
emphasis on their use for the recording and distribution of live
music, as done by the followers of the Grateful Dead, Phish, and
others.
To subscribe to DAT-heads, send a message to:
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dat-heads-request@virginia.edu
with "subscribe" as the message.
**********************************************************************
Q4.1.4 What are some related WWW (World-Wide Web) pages?
Blurbs below (in no particular order):
Ram's music making tips
Mixmasters (see also previous question)
Shaolin Temple
Internet Showcase
La Factoria Del Ritmo (The Rhythm Factory)
The Buddy Project
**********************************************************************
me@ram.org (Ram Samudrala) writes:
Check out my recording tips page:
http://www.ram.org/music/making/tips/DiY.html
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Everything in the tips was done using a 4-track...
**********************************************************************
Mixmasters is a worldwide group of people who exchange audio tapes and
advice for the sole purpose of helping improve each others' production
values.
We now have a home page! Start at
http://homerecording.com/mixmasters.html
**********************************************************************
haibachi@tiamat.umd.umich.edu (Jay Itchon) writes:
well to tell yo why this fits on this group almost all of the music
was recorded on a 4 track! Hello there! theres a new homepage for
the label Shaolin Temple its being worked on but check it out anyway!
http://www.umd.umich.edu/~eth
write me at haibachi@umd.umich.edu
Shaolin Temple is home to Noise, PUNK, Hardcore, Noise and more noise stuff.
Haibachi/Jeenifer Wolski/Ameyeseekayeewhy?/Latch/Friendship/Dreams/...
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**********************************************************************
crick@ccnet.com (TEFKAR) writes:
Internet Showcase URL
We have set up a homepage to showcase various local talent, with song samples,
bios, etc... at http://www.ccnet.com/~showcase/
Feel free to stop by and check out some of the local Bay Area talent.
BTW, the site is currently under construction, so please pardon our mess:)
for info, contact showcase@ccnet.com
**********************************************************************
LA FACTORIA DEL RITMO (THE RHYTHM FACTORY)
The first musical fanzine on the INTERNET about Spanish Music.
[As far as I can tell, it's only in Spanish. Esta bien, por que me
interesa leer espanol, aunque no hablo ni escrito muy bien. --DSF]
LA FACTORIA without pictures:
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http://www1.uniovi.es/musica/
LA FACTORIA with pictures:
http://www1.uniovi.es/musica2/
e-mail contact address: zz93f010@polar.etsiig.uniovi.es
(Perez Sanchez, Florian Manuel)
**********************************************************************
"The Buddy Project is an electronic clearing house for musical
creations by both electronic and acoustic musicians as well as
vocalists and hey, even beat poets, if you'd like."
URL: http://www.buddy.org/
buddymeister: tnolan@frymulti.com (Timothy M. Nolan)
**********************************************************************
Q4.1.5 What are some related ftp sites?
A great ftp site for computer audio formats and utilities is Guido van
Rossum's place:
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ftp://ftp.cwi.nl/pub/audio
The above directory has a hypertext "index.html" file.
S4.2. Information not online
Q4.2.1 What are some related magazines?
1. GAJOOB Magazine, published by Bryan F. Baker. This is the older
sibling of the GAJOOB online "DiY report".
"GAJOOB Magazine has been in publication for nearly eight years and
has a small (2000+), but devoted international readership. Due to the
success and effectiveness of the elctronic newsletter, GAJOOB
Magazine's DiY Report, the focus of GAJOOB Magazine itself has
recently evolved to be a more in-depth look at DiY recording as a
craft and DiY recording artists as creative artists at work. Future
issues of GAJOOB Magazine will focus less on contact information and
reviews and more on people, issues and DiY recording as a craft,
leaving the DiY Report newsletter to keep DiY recording artists
up-to-date with essential information on the most current DiY
activities."
CONTACTING GAJOOB
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Direct any further questions, comments or job offers to:
E-MAIL: gajoob@utw.com (preferred) or gajoob@aol.com
Postal enquiries, submissions, Baby Ruth candy bars and the perfect cup of
coffee (or the perfect coffee cup) can be sent to:
GAJOOB Magazine, PO Box 3201, Salt Lake City, Utah 84110.
2. IMPROVIJAZZATION NATION. A writeup from GAJOOB DiY:
ZZAJ PRODUCTIONS " . . . publishes (e-mail) periodic "updates" on new
releases, recording activity at our "In The Shadow" studios & various other
info regarding D.I.Y. activities throughout the world. If you "do" I.Y., or
know of folks who do, drop me an e-mail. IF you want the list, please
specifically ASK for it in your e-mail. We also have information about our
'zine, IMPROVIJAZZATION NATION, which reviews D.I.Y. tapes of ALL genres,
as well as some limited, short pieces of poetry. Let us HEAR from you, ok?"
[Rotcod Zzaj, Editor, IMPROVIJAZZATION NATION 'zine, & perpetrator of Zzaj
Productions; SNAIL: 5308 65TH Ave SE, Lacey, WA 98513; EMAIL:
rotcod@halcyon.com]
3. Home & Studio Recording - ??
4. Sound on Sound - A home recording magazine in the United Kingdom.
benl@mojo.europe.dg.com (Ben Last) writes:
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Sound on Sound deals with all aspect of home recording (with a bias
towards synths), but includes samplers, sample cd reviews, demo tapes
reviews, 'how to' articles, etc. Thoroughly recommended!
Q4.2.2 What are some related books?
1. The book that comes with your 4-track machine!
2. The Musician's Home Recording Handbook, by Ted Greenwald.
kmk@cornell.edu (Jason Kelsey using the account of Kristina Kelsey)
writes,
"I found it to be a very good source of information for beginners and
intermediates."
3. Home Recording Techniques, by J.D. Sharp.
longmire@mcs.com (Bob Longmire) writes,
It's published as an Alfred Handy Guide, by Alfred Publishing Company.
I found this is a music store, but the publisher's address on the back
cover is:
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Alfred Publishing Co., Inc.
16380 Roscoe Blvd.
P. O. Box. 10003
Van Nuys, CA 91410-0003
4. New Ears: The Audio Career & Education Handbook
mwdrews@mailbox.syr.edu (Mark Drews) writes:
For those of you considering a career in the wonderful world of music
recording or audio engineering .. The guide lists most everything
you'd ever want to know about music recording programs and
schools. Drop me your email address, and I'll forward a press release.
Happy tracking!
Mark Drews
Senior Audio Engineer/Instructor
SU School of Music
5. Cameo Dictionary of Creative Audio Terms [what is this?? --DSF]
6. Hot Tips for the Home Recording Studio, by Hank Linderman, Writer's
Digest Books, 1994. 161 pages. ISBN 0-89879-651-2.
This slim volume is a collection of thoughts from a producer of
hundreds of demo tapes in the L.A. area. I found it extremely
well-written and up-to-date; it does not try to cover everything. It
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has an index.
From the back cover: "[Hank] helps you lay down the music the way you
want it to sound, with advice for: getting the best from your mixer,
multitrack, compressor and other gear; arranging your song; keeping
life in your music; unlocking the mysteries of time code; preventing
acoustic and electrical noise from interfering; laying down good
tracks, using 'production prosody,' 'ramping up'; 'punching in';
working with singers; mixing; .."
"For a FREE catalog of all Writer's Digest Books, write Writer's
Digest Books, 1507 Dana Avenue, Cincinnati, Ohio, 54207, or call
1-800-289-0963." They have books on lyrics, poetry, melody, and other
related issues.
7. The Musician's Guide to Home Recording, by Peter McIan and Larry
Wichman, Simon & Schuster, Inc, 1988. 285 pages. ISBN 0-671-60189-X
PBK, or 0-671-65754-2.
This tome is written by the producer of Men At Work and Mr. Mister,
with help from a free-lance writer. It covers a lot of ground, and is
a little dated. It has an index, bibliography, and glossary.
From the back cover: "Learn: how to specifically EQ and recod
synthesizers, drums, bass, strings, guitar, brass and piano; how to
reduce tape noise and generation loss during a bounce; how to
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e-create sounds you hear on hit records; how to properly apply reverb
and special-effect echoes; how to perform overdubs; how to test room
acoustics; how to achieve an effective blend; .."
8. Music Through MIDI: Using MIDI to create your own electronic music
system, by Michael Boom, Microsoft Press, 1987. 271 pages, with
glossary and index. ISBN 1-55615-026-1.
I have not used the book much, but it looks like a gentle yet complete
introduction to MIDI.
Chapter 1 MIDI: An Overture, Ch. 2 Sound and Music, Ch. 3 Synthesizing
Sound, Ch. 4 MIDI Connections, Ch. 5 MIDI Messages, Ch. 6 Computers
and MIDI, Ch. 7 Real MIDI Equipment, Ch. 8 Computerized MIDI Systems,
Ch. 9 MIDI in Live Performance, Ch. 10 MIDI in the Recording Studio,
Ch. 11 MIDI in Education, Ch. 12 MIDI at Home, Appendix A Getting
Technical, Appendix B Companies Mentioned, Appendix C Further
Information.
9. Recording Demo Tapes at Home, Bruce Bartlett, 1988.
I have no info on whether this is a good book or not.
Q4.2.3 What are some related videos?
1. More Top Secret Home Recording Techniques, Vol. 3, Curt Miller
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Productions, 1993.
What an excellent idea! Illustrating recording techniques on
videotape, where you can see what they're doing and hear the
difference it makes.
This tape covers: parametric EQ, reverb and delay, noise gates,
monitoring, "advanced mixing."
Curt Miller Productions' address: 4238 Childress, Houston, TX 77005.
Phone # (713) 665-3135.
S4.3. Company information
Q4.3.1 What is Tascam's address and phone #?
Tascam Professional Division
TEAC AMERICA inc.
7733 Telegraph Rd
Montebello, CA 90640-9978
http://www.tascam.com
Phone 213-726-0303 (ext. 617 for product support)
FAX 213-727-7635
Note: they seem unresponsive to email, but calling them might find you
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that old manual you've been looking for.
Q4.3.2 What is Musician's Friend?
A mail-order company that can be found at http://www.musiciansfriend.com
Phone # 800-776-5173
Studio FAQ v0.1
(c) 1996 Hal Haygood -- see answer 0.5 for details
If you got here from Stomp Across the World Wide Web, you can get back by clicking here.
TABLE OF CONTENTS
Section 0: Meta-FAQ
• 0.1) About this FAQ
• 0.2) FAQ locations
• 0.3) About the author
• 0.4) About the author's band
• 0.5) Disclaimer and Copyright
Section I: General
• 1.1) Why should my band record something?
• 1.2) Why should my band record in a studio?
• 1.3) What is the general process involved in recording?
• 1.4) What are the pros and cons of recording a live show?
Section II: Deciding
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• 2.1) How much does it cost?
• 2.2) How long does recording take?
• 2.3) How do I pick a studio?
• 2.4) I've heard that recording breaks bands up.
• 2.5) Do we need a producer?
• 2.6) Where can we get the money?
Section III: Before you go
• 3.1) What songs should we record?
• 3.2) Should we record covers?
• 3.3) What gear should we take?
• 3.4) How should we prepare?
Section IV: In the studio
• 4.1) What is the recording process?
• 4.2) What are basic do's and dont's?
• 4.3) I'm getting SO sick of this song.
• 4.4) I've played this part about a zillion times now and still can't get it.
• 4.5) Why is everyone else screwing up so much?
• 4.6) No matter how well I play this, everyone wants me to do it again.
• 4.7) I'm hungry (Thirsty, nic-fitting, sobering up).
Section V: Mixing
• 5.1) What is the mixing process?
• 5.2) What are the basic do's and dont's?
• 5.3) If I hear this song one more time, I'm going postal.
• 5.4) I want more me in the mix.
• 5.5) I don't like what the producer's doing.
Section VI: Reproduction
• 6.1) Should we do tapes, or CD's?
• 6.2) Should we master it?
• 6.3) How much does it cost?
• 6.4) Should we do a slick package, or handwritten labels?
• 6.5) How long does it take?
SECTION 0: META-FAQ
0.1) About this FAQ
This is version 0.1 of the Studio FAQ, aimed at bands looking at their first studio experience.
There's not much here for bands with lots of studio time, but you're welcome to check it out
anyways.
If you have information you feel should be included, email me at hal@suncap.com for inclusion in
the next version. Actually, I'd appreciate any feedback at all, because I love the attention and this is
the first time I've done anything like this.
I did this because I think Harmony Central's the coolest thing since the tube amp.
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This version was completed 12/4/96.
0.2) FAQ locations
This faq can be found at http://www.haydensferry.com/stomp/studio-faq-01.html
0.3) About the author
The author of this FAQ is Hal Haygood, lead guitarist for Stomp Gospel (see 0.4). In addition to
my fantastic guitar playing (hell, at least *I* think it is), I'm an in-house and on-site engineer for a
systems integrator. Oh, and a full-time student working on a BS in CIS. I'm a tube nut. I play a
Telecaster through a SF Bassman. I'm in Phoenix. I can be reached at hal@suncap.com.
0.4) About the author's band
Stomp Gospel is a roots-rock/alternative band from Tempe, Arizona. We consist of Charlie Curtis
(vocals), Chris Connelly (bass), Bob Miller (drums), Paul Fredricksen (rhythm guitar) and me.
We've recorded a five-song demo and a CD project at Blue Sky Recording Studios in Tempe.
We're signed to Hayden's Ferry Records.
0.5) Disclaimer and Copyright
First and foremost, keep in mind that this is all being tainted by my experiences in the studio. The
band didn't help me with this (I didn't ask). Therefore, there's probably a slight emphasis on guitar
work and smaller studios. I don't know how things work at the really expensive studios; Blue Sky
is a fantastic place to record and it's pretty inexpensive.
I made up the questions myself. If you have other questions or answers, or if you think I'm totally
wrong, please email me at hal@suncap.com for inclusion in the next version.
There are lots and lots of my personal opinions in here. Your mileage may vary.
The information contained in this document is for informational purposes only, and the reader
assumes all responsibility for damage resulting from following the suggestions made.
This document is copyright 1996 to Hal Haygood. Redistribution in electronic form is encouraged
as long as this copyright message is included and no fee is charged other than standard online
access fees or media fees. Please contact the author for rights to redistribute this document in print
form and/or for a fee. All rights not specifically granted in this paragraph are reserved to the
author.
If you link to this document, I'd appreciate it if you let me know at hal@suncap.com.
SECTION I: GENERAL
1.1) Why should my band record something?
Lots of reasons. First, and perhaps most important, you'll need something to give to clubs for
booking. If you don't have a tape, most clubs won't even see you. Another reason is to shop labels;
even though, realistically, the chances are slim, some bands have gotten contracts from demo
tapes. You might want something to prove to your parents that you're doing more than just making
noise in the garage. You might want something to sell or give to your friends.
Chances are, if you're reading this, you should be recording. The author's personal opinion is that
recording is a GOOD thing. There are several benefits to the recording process other than the
finished product -- better musicianship, a chance to hear your songs without playing them, a
chance to maximize your full potential as a band, a chance to make your music something more
than it is so far.
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Finally, it just feels good. It's validating: "Yes, I really AM a musician, not just some boob
with a guitar."
1.2) Why should my band record in a studio?
One word: QUALITY. Even the worst studio can give you a product that is FAR better than a
four-track. Mind you, I'm not disrespecting the shoestring demo; almost every band records a
practice and shops clubs with the tape. In fact, recording practices is a great way to prepare for the
studio.
If you just want something to practice against, don't go to a studio; it's too expensive for something
you're only going to practice your wicked licks with. But if anyone else is going to listen to your
music, it needs to be something that doesn't have room noise, airplanes flying over (Led Zeppelin's
Black Country Woman notwithstanding), noise, and muddy drums.
If you haven't been to a studio, you may not be aware of the things that they can do with sound --
it's pretty amazing.
1.3) What is the general process involved in recording?
There are 7 stages: Deciding, Preparing, Recording, Overdubbing, Mixing, Mastering, and
Reproducing. (If you're with a label or have high aspirations, add Distribution at the end.)
Deciding: This is where you pick a format (CD/Tape), a length (two-song demo, five-song EP,
nine- to fourteen song project, twenty- to thirty- song double CD), songs, a studio, possibly a
producer and/or engineer, and a duplicator.
Preparing: This is where you practice, over and over and over again. Then you get together your
recording money, get your gear in shape, and book time. Then practice some more.
Recording: This is the time you actually spend in the soundproofed room, crooning, picking,
slapping, shaking, and doing that thing you do. Compared with the time invested in everything
else, it's woefully short. It's also the most fun.
Mixing: Once everything is tracked (put down on master tape), it needs to be prepared for
mastering. Everything is made to sound just right, noise is subtracted, reverb is added, and
everything ends up on a DAT.
Mastering: This is an optional (author's opinion: mandatory) step where engineers at the
reproducer run your tape through the appropriate filter for the format (CD, vinyl, or tape). The
dynamic range may be compressed or expanded, the whole mix is EQ'd, and everything ends up on
another medium. This may be DAT or hard disk (for CD, vinyl, or tape), or 1/4" tape (for
cassettes).
Reproducing: For CD's, they take the final master tape/disk and create a "glass master", the glass
disc that is the mirror-image of your CD. This is used to press the plastic for the CD. Cassettes are
reproduced by bin-loop, where they take the 1/4" master tape, make several copies, string them
together, and use that to feed a cassette-duplication machine. I'm not sure how vinyl is pressed, but
you can ask any reproducer how it's done.
1.4) What are the pros and cons of recording a live show?
Pros: You can capture the energy of a live show. Some bands find it difficult to take the live
energy and reproduce it in the studio. You can also capture the audience response. There's less
recording time involved, only a few hours on-site.
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Cons: It can be VERY expensive. If you're already running everything through a board (even
all the drums), that mitigates the expense somewhat, but you still need to pay an engineer, pay for
the recording equipment, negotiate with the club, and the like . If you're just setting up mikes in the
audience, beware that your tape will be much lower quality than a studio recording.
Bottom line (author's opinion): Save it for when you've got a label to pay for and arrange it.
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SECTION II: DECIDING
2.1) How much does it cost?
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Studio time is fairly expensive. For a first project, figure on $30 - $80 per hour spent recording
and mixing, plus mastering and duplication costs. Don't forget the cost of J-cards or CD inserts.
Some studios offer package plans; the author has seen some in the Phoenix area for $2000 that
include a 1-week studio blockout and 1000 CD's with 2-page inserts. Make sure you understand
EXACTLY what that cost includes and don't sign ANY contract without taking it to a lawyer.
Don't give up the rights to your music unless you're sure you know what you're doing.
2.2) How long does recording take?
That's really, really variable. The author's band did a five-song demo tape in two days
(recording/mixing) and 3 weeks (printing and reproduction). However, we're now working on a
much more involved project that has taken just under a year so far with six months left to go.
There are lots of factors involved. How serious of a project is it? Is it something to go in record
stores, or to get gigs? Are you making CD's? (CD's take longer to reproduce.) How much of a
perfectionist are you? How competent is the engineer?
2.3) How do I pick a studio?
(This is my experience. Your mileage may vary.)
• 1) Ask your friends. They may have had good or bad experiences.
• 2) Look in the newspaper, particularly independent weeklies or music rags.
• 3) Pick a price you're willing to spend per hour.
• 4) Call all the studios you found in 1 & 2 that fit your price range.
• 5) If they're rude to you on the phone, find out if it's the owner. If it is, don't bother. If it's
just an unruly engineer, ask to talk to the owner. You should start whittling your list down
here.
• 6) Arrange to visit the studios left on your list, with the whole band if possible. See what
their facilities are like, what kind of equipment they have, how friendly the employees are,
and most importantly, how comfortable you feel there. If you're not comfortable, you can't
make music to the best of your ability.
• 7) If, after the visits, there are more than one left, call them and tell them that. Mention that
you're looking at a couple of places, and you like them the best. See if they'll cut a deal. (If
you're not good with negotiating, be careful not to piss them off.)
2.4) I've heard that recording breaks bands up.
This can be true, but it's a matter of attitude. Recording does lots of things to the structure of the
band, and introduces new stresses that are unfamiliar. The pressure is high, and it's tempting to be
a "rock star" about things. (I don't recommend it.) Egos can run wild, and nerves can fray.
Really, though, recording is a test of a band. If you can't stand to work together on something you
all agree on, how will you handle living with everybody on tour for 9 months?
One of the most surprising things for the author is what happened AFTER the recording. Once
something is down on tape, and "done," it's human nature to reflect on everything that's led up to
that point. Practices may not seem as fun, and it may feel like there's something missing. I felt like
we were plateauing badly. The solution to that? Two things: write more songs, and record more. :)
2.5) Do we need a producer?
Yes and no. Your band should be the most important producer; after all, it's your music. However,
it's really handy to have another pair of ears around that's NOT emotionally involved. To me, a
producer is a lot like a proofreader and editor. They take what you do, tweak it a little or a lot,
point out mistakes you might not have noticed, and put it all together.
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It's generally a bad idea to have a band member's significant other (boyfriend, girlfriend,
spouse, dog, parakeet) as a producer unless they're a professional.
2.6) Where can we get the money?
Rich relatives. :) Seriously, it's handy to have someone to borrow from if you're not making money
gigging. Make sure you pay them back as soon as you can. Save some.
SECTION III: BEFORE YOU GO
3.1) What songs should we record?
Record your best songs. It's important to make a distinction between good live songs and good
recorded songs, though; some of the most powerful live music doesn't translate well to a CD
without the audience to feed off of.
It's better to record fewer, better songs than to record more, crappier songs. You may not want to
admit it, but we ALL write crappy songs (some more than others).
Record songs that are fun to play. You're probably gonna end up playing them a lot.
Record a variety of songs. If you've got some hip-hop, some metal, some pop, and some country
songs, maybe pick the best songs of each type. Critics are very unfriendly to bands whose recorded
music "all sounds the same."
3.2) Should we record covers?
Probably not. Two definite no-no's are to record covers that are still under copyright (you may do a
better version of "Panama" than Van Halen themselves, but that doesn't make it legal) or to record
songs that everyone and their sister has already covered ("Mustang Sally," "Little Wing," etc.)
3.3) What gear should we take?
As little as you can get away with and still get your sound.
For guitarists, I'd suggest an amp (head and cab), a guitar, your basic effects, and enough cables to
put everything together. It's been my experience that studios have spares for most stuff that can
fail. They probably also have better effects than your Crap-o-Tron flanger, too. Bring spare tubes,
though.
Bass, for a demo project, is usually recorded directly into the board, so there's no need to bring an
amp.
Drums can sometimes be provided by the studio, unless you're so totally in love with your kit that
you can't play anything else.
Singers should bring a big bottle of whisky. (Just kidding.)
3.4) How should we prepare?
Let me preface this by again reminding the reader that this is based on my experience, and that
every band has its own style and dynamic. This worked for us; it may not work for you.
The most important preparation to do is of the music. You're paying for studio time; it's cheaper to
make sure everything's ready beforehand. Practice time is free, or close to it. Use it. Practice the
songs over and over, until everyone could play their part backwards and in their sleep.
One thing that worked for us was deconstruction. We'd play a song a couple of times, then break it
down into parts. First, just drums and rhythm guitar, then drums and bass, then rhythm guitar and
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lead guitar, then rhythm guitar and bass, etc. After everyone was set, we'd bring everyone back
in and play it a couple more times.
If there are going to be any changes to the arrangement, sounds, or lyrics, now is the time to make
them. (Speaking in general terms, of course; being in the studio gives a new atmosphere that
naturally changes the music somewhat.)
Get your gear in shape. New strings, drum heads, tubes, or anything else you might need. Get the
guitars and bass set up. Get the piano tuned. Fix the zither. (What exactly is a zither, anyways?)
SECTION IV: IN THE STUDIO
4.1) What is the recording process?
There are a few general processes. Some bands prefer to track everything live, with the whole band
in the studio. This helps capture the interplay of musicianship better, but it can make for looooong
times in the studio and many, many takes. Most studios have a drum booth, which allows for live
drum tracking, and closets or booths for guitarists and amps.
Most bands take the incremental route. The most important track (several tracks, actually) is the
drum track, so it's imperative that it be flawless. Doing it as a "build-up," the drum track is tackled
first, either with one or two other musicians or the whole band playing along. Once the drum track
is perfect (or acceptable, depending on the budget), the bass track is laid down to complete the
rhythm section, and the other tracks are overdubbed, building up the finished track. Generally,
vocals are laid down as the last major track, then atmospherics like tambourine or backing vocals.
Electric guitars are miked at the speaker, or run through a direct box. (A direct box takes a guitar
signal or an amp's speaker output and bring it up to the level of the mixing board.) Same gig for
bass. Acoustic instruments are miked in a soundproof booth.
The mikes, or direct boxes, are run to a big mixing board, 16, 24, or 32 channels wide, maybe
more for bigger studios.
The engineer handles the "monitor mix" -- the mix used for the monitor headphones. It's not the
same mix that's actually laid down to tape, however; that's mixed according to the dynamic range
of each instrument. Monitoring is generally done with headphones, and the mix is also piped
through speakers in the control booth.
There are two kinds of recording, analog and digital. Analog is laid down on big (1" or 1-1/2")
tape, which records 16 or 24 channels. Digital is laid down on similar tape (for the expensive
studios) or ADAT. ADAT (Alesis Digital Audio Tape) is recorded on special VHS videocassettes
that hold 8 tracks of CD-quality digital audio. Both can be used simultaneously, if they're
synchronized. They get synchronized with SMPTE, a time-coding scheme that gets recorded on
one analog track. (SMPTE stands for the Society of Motion Picture Technicians and Engineers, I
think. It's pronouced "simpte.")
Expect to have to overdub lots of things. Remember, this is your chance to get it right; you'll be
hearing that one flubbed note over and over again. (See the "Mixing" section for more on that. :)
4.2) What are basic do's and dont's?
Don't mistreat the studio's equipment.
Don't yell at someone for screwing up.
Don't expect to make it perfect on the first try.
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Don't waste a lot of time; in this case, time really IS money.
Don't be too wasted to play or sing. A buzz might be okay, but make sure that you really want to
capture that buzz, 'cause you will.
If you're unsure, don't settle unless money is really an issue.
Do have backup instruments.
Do be nice to the engineer.
Do bring snacks and water.
Do avoid dehydrating drinks (beer, coffee, Coke) if you're singing.
Most importantly, do observe the rules that the studio has in place. Most studios are nonsmoking,
so you'll just have to smoke outside on breaks. If there's a big sign in the control booth that says
"No Drinks on the Console", pay attention. Mixing consoles are expensive, and you don't want to
have to pay for that too. Drugs (even pot) are generally a no-no, whether or not there's an explicit
rule in place.
4.3) I'm getting SO sick of this song.
Yup. So is everyone else, probably. Take a little break, have a smoke, do whatever. Maybe move
on to something else for a little bit. There's not much more advice I can give here; it happens, it
sucks.
4.4) I've played this part about a zillion times and I still can't get it.
Yup. It happens, it sucks. There are a couple of options here. You can take a break, go scream at
stuff, try to relax a little, or get a pep talk. Or you can try to rewrite the part to something simpler.
Sometimes simpler things, played well, come off LOTS better than complex things, played poorly.
It's disheartening to acknowledge limitations, but it might be for the best.
4.5) Why is everyone else screwing up so much?
The same reasons that you are. Nerves, pressure, fatigue (physical AND mental). Relax a little. If
you're not stepping on sexual mores, backrubs work well. Maybe have a significant other around
for it. :) One of the best ways to deal with this is positive feedback -- "strokes." Let the person
who's having a tough time know that you believe in their ability, and you're not frustrated with
them for screwing up. Even if you are. Keep in mind that when it's your turn, you're gonna screw
up just as much as them. It's a part of the process, not something to be upset with.
4.6) No matter how well I play this, everyone wants me to do it again.
They know where your potential is, or at least they think they do. If you're happy with a take, and
you don't think you can do it any better, let 'em know. If you're really happy with it, stick to your
guns. (This is where a producer can be either an angel or a demon. If they like it, they're an angel.
If they're not happy with anything you do, they're a demon. An uninvolved set of ears, however,
can give really good feedback.)
If nothing else, you can always overdub it again later. Chances are, after hearing it a bunch of
times the way you want it, they'll come to accept it as part of the song.
4.7) I'm hungry (Thirsty, nic-fitting, sobering up).
Eat. (Drink. Smoke. Drink.) You can't play or sing up to your potential if you're uncomfortable. If
tension is too high to suggest it, wait a few minutes, but get it done.
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SECTION V: MIXING
5.1) What is the mixing process?
Basically, you sit in a booth while an engineer tries a bunch of different things with the mix.
A song is mixed the same way it was recorded; drums first, then bass, then everything else, with
vocals last. Each track gets its particular sound down, effects are added at the board, and general
levels are set. Once the sound of each track is dialed in, the whole song is run through to listen for
any places where special mixing attention is needed.
Once everything's set, the mixing engineer does a "dry run", running the entire song through with
all the settings and all the tweaks. Finally, the DAT is positioned and the engineer runs it for the
final mix.
Some studios have automated mixing consoles that handle the tweaks with no intervention from
the engineer. (I think these are really keen. You get to watch the sliders and knobs move by
themselves!)
5.2) What are the basic do's and dont's?
Do voice your opinion if you don't like something.
Don't touch the board. Leave that to the engineer.
Do try to be patient. (See 5.3).
Don't get burned out on one song.
5.3) If I hear this song one more time, I'm going postal.
Yup. It happens, it sucks. Part of the process. Mixing is arguably the most important step in the
recording process, and it's very, very important to stay focused. Unfortunately, mixing also
involves listening to the same song over and over and over again. If it feels like you're losing touch
with the song, or losing focus, take a break or move on to another song for a while.
5.4) I want more me in the mix.
Before you say anything, make sure you really do. Try to picture the song as a whole, not just a
collection of solo pieces. If you've carefully considered the interplay of all the parts, and you still
think that your solo should be louder, say something. Be prepared to be shot down by someone,
and remember, if you've agreed to have a producer, make sure you're not gonna piss 'em off. (See
5.5)
5.5) I don't like what the producer's doing.
Mmm, this is sticky. First off, the producer should be open to your ideas. They're your songs. If the
producer isn't paying attention to anything you say, you might want a new producer. Talk to him or
her. Try not to get emotional about it; explain yourself rationally. If nothing else, try for a
compromise.
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SECTION VI: REPRODUCTION
6.1) Should we do tapes or CD's?
Yes. :)
My guidelines: For a first project, a demo, anything under 9 songs, or a shoestring, do tapes. For
everthing else, if you can afford it, do CD's. They carry more prestige, last longer, and sound
better.
6.2) Should we master it?
Yes, unless you REALLY can't afford it. Mastering makes a difference that's not so much audible
but emotional. Well, it's audible, too, but it's only really noticeable in its absence. Remember that
every CD you see on your shelf is mastered. (Strangely, most of them by one man -- Bob Ludwig
at Masterdisk in New York City.)
6.3) How much does it cost?
These prices are estimates based on memory. Your mileage may vary. Check with your duplicator
for real numbers.
A run of 100 tapes costs about $250 from a reputable, quality duplicator. J-cards are extra.
The glass master for a CD costs about $500, less if you're getting duplication from the same
company. CDs cost $1-$2 apiece, depending on the quality. Jewel cases, inserts, and backcards are
extra.
Most duplicators offer package deals where they'll master, duplicate, and provide printed material
for one price. Most include shrink-wrapping.
If you'll be distributing your product in record stores, you'll need a UPC code. I understand that
they cost about $300 to register, then you need to pay for them to be printed. (Or you can include it
on the backcard or J-card.)
6.4) Should we do a slick package, or handwritten labels?
For a gig demo, handwriting is fine. For anything else, you should really have printed labels and
cards.
6.5) How long does it take?
Most tape duplicators take about a week, more for runs over 2500. CD duplicators take from 3
weeks to 3 months, depending on their backlog. Backlogs increase late in the year, for some reason
-- maybe Christmas compilations. They ease up around February, and late spring is the fastest time
of the year.
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