Conference napalm::commusic_v1

    
    
    The best way to avoid tape saturation and still get a decent recording
    level is to get a compressor.  Its one of the basic necessities
    of life in multitrack recording.  And you'll constantly find new
    uses for one too.
    
    ralph
    
    

Maybe the singer is just singing too loud?  No really.  I've seen singers
on TV who practically swallow the microphone; they should be 6 inches to
one foot away.  Then the singers have to hear themselves in headphones,
balanced against the rest of what is going on.  They will unconsciously
change the level at which they sing according to what they hear, so set
the mix first without them able to hear it, then adjust their monitor level.
If they hear the level change, THEY will change too, in the opposite direction.

Compressors and limiters are often used on mic inputs to protect against
overloads, but they are really making up for poor control on the part of
the vocalist.

    
    Hmmmm... I'll give that a shot Paul.  I have been close to the mike
    and have been setting the levels with the headphones on.  Hmmm...
    
    Thanks,
    
    Ben.
    

    One thing that occurs to me is that you shouldn't necessarily
    "believe" the needles.  One of the first things I did when I got
    my 4-track was to make the following experiment.
    
    I cranked up fairly level (dynamicly speaking) sequence on my ESQ-1
    and had it go into one track.  On another track I had a live mic.
    
    I had the sequence repeat indefinitely but after each iteration
    I would raise the volume one notch on the recorders meters and recite
    the meter reading into the mic so I could hear what level it was
    recorded at on playback.  (It turns out that the meters on my unit
    do NOT display the same data on playback as they did during record.)
    
    I found that I could 'overdrive' the meters well above the 0db
    standard.  
    
    I later talked with a recording engineer who sorta said that the
    optimal levels are a function of the machine itself, the calibration of
    the machine, and the grade of tape used.   It was also satisfying
    when he pointed out that the grade of tape I was using recommended 
    using a higher level right on the packaging (which of course, I
    had never read).
    
    	db

    re: -1  I don't take the meters as gospel but they are a good 
    guide and in the same trick the ears say "wow...  thats bad!".

    
    To try to keep my pathetic vocals within th emeters range, I wen
    t ahead and  bought a Shure SM10...the headset mic.  This way I'm
    not eating the mic, and I'm always the same distance away, so it
    has taught me to try to sing the same volume for most every song.
    
    It is also great live....the only drawback is that there is no cough
    button, or any ohter way to keep your heavy breathing off the mix
    other than to move the arm.
    
    Ashley

    One more vote for, I mean against, 'eating' the mic.. the famed
    (all together now) Fletcher-Munson 'proximity' effect.. bass goes
    WAY up the closer you get.. I considered fastening a sharpened pencil
    to the microphone to get one vocalist to back off.. 

    Or is the Fletcher-Munson curve that says the farther AWAY it is
    the less highs arrive... I forget.
    
    A comment on:    
    
>< Note 1158.4 by DREGS::BLICKSTEIN "Dave" >
>    One thing that occurs to me is that you shouldn't necessarily
>    "believe" the needles.  ...     (It turns out that the meters 
>    on my unit do NOT display the same data on playback as they did 
>    during record.)

    A well adjusted recording unit will ALWAYS show the SAME LEVEL
    on playback that was showing during record. This is a function
    of the input level adjustment and the tape bias setting.. 
    your deck should be biassed for the ONE type of tape you use...
    Cassette decks with their limited speed are even more prone
    to level and bias-related garfing than open-reel decks.     
    
    karl
    

    This thing about not 'eating' the mike, is this a difference between
    live and recording? When I do sound for live bands, I find that
    if the vocalist doesn't eat the mike, they're constantly changing
    the distance between there mouth and the mike, so I've always
    recomended that the mike be eaten, then I adjust the EQ to keep
    the bass reasonable. I've not really had very much recording
    experience, but logic would tell me that I'd face the same problems.
    I've also found that my mikes (AKG-321's) have a very thin sound
    at 6 inches.
    
    pcv
    

    Proximity effects are always much more a problem for live work than
    studio work, primarily due to the fact that during studio sessions,
    the vocalist usually maintains the same position from the mike...or
    at least has no excuse for not maintaining the same position.
    
    In live work (of course, especially with hard drivin' raunch'n'roll),
    movement is quite often a necessary stage expression.  You know,
    bending backwards doing that crazed riff, then coming back up to
    the mike for the next verse, etc.
    
    I, too, have used Peter's approach in handling live mike work: set
    the EQ/level for worst-case conditions, then agree with the vocalist
    to eat the mike.

    I tried the pencil taped to the mike routine last night.  I took
    the liberty of using the *eraser* side toward me instead of the
    sharpened end.  That really seemed to help though coming
    up with an adjustment between the -10 and 0 (i think dB -- I'm still
    new at this) on my meters without peaking at times was still a fairly
    fine line.  I found most of the time when I would end up above 0
    or into the peak range was a blast of short duration.  Thinking
    of microphones which have screens on them I took a kitchen towel
    and covered the mike.  This made for less attacks on the peak LED
    but brought the signal strength down some.  Upon boosting the input
    strength I started going back above 0 and peaking.  
    
    There must be a fine line here somewhere; I think if I keep on
    trying I'll find it.  
                         
    Ben.

    Unless your hellbent on a lo-tech, lo-bux solution, why not bite
    the bullet for a comp-limiter? This is the kind of thing they 
    were designed for...
    
    Admittedly, funnin' around with towels, erasers, toilet paper rolls,
    and newt eyes can be rewarding and educational, but if you wanna
    sound like 'the big kids' you gotta have the toys...
    
    Edd

    Your right Edd, and I figure on it adventually.  I'm just trying
    to understand, use, and pay for what I have already first.  
    
    How long my patience lasts will probably be the ultimate deciding
    factor...
    
    Ben.

    
    Yep Edd's right the only real answer is a compressor. Check out
    the Alesis micro rack  verions cheep and stereo and good.
    
    				Paul.

You should be using a mic with a "pop screen" on it.  This will not do anything
about times when you sing too loud, but it will prevent puffs of air from
overloading things.   Something else to try is to not sing directly at the
microphone, so your breath is not hitting it.

I have heard of some technicians putting up two microphones, a placebo for
the vocalist to swallow, and another one one foot behind.  Obviously this
only works with fixed mics.  For hand mics you just gotta learn where to
hold it and keep your arm locked as you sway around.  I also read about
one singer who got his shoes nailed to the floor.

When I have seen people recording in a studio, they are almost always a
foot OR MORE from a fixed microphone.  It may be that professional singers
can sing much louder than you can, so can be farther away.  (Opera singers
rarely need PA systems to be heard in the last row)  If you try to sing
that loud without voice training, you could hurt yourself.

    re .7 - the proximity effect is not the Fletcher Munson effect.
    The latter is a psychoacoustic effect that results in our perceiving
    less bass at low absolute volumes.  This is what the "loudness
    componsation" switch on a hifi amp is for.
    
    The proximity effect is a physical effect that results in microphones
    exhibiting ehnhanced low frewuency response when close to the source.
    I don't know if it's named after anybody (Fred Proximity?  He invented
    the fuse too, didn't he?).
    
    len (can I correct errors other than spelling and vocabulary?)
    

    Is the actual frequency spectrum effected at all by distance??
    
    Seems to me that low frequencies carry farther through any given
    medium (Air, wood, metal, pea soup) with high frequencies dropping
    off...
    
    ...in which case only a high-end roll-off would be needed to suggest
    increasing distance. 
    
    I stand corrected. (Thanks to my Dr. Scholl's!!)
    
    Edd

    No, the proximity effect is more than just the differential absorption
    of high frequencies vs. low frequencies.  The bass enhancement is
    quite a bit more than you'd expect from such differential rolloff.
    There's an actual bass boost rather than an apparent lack of bass
    loss.
    
    Uhm, is this the note about spatial relationships?  If it is, the
    latest EM has an interesting article on the subject.
    
    And, uh, Edd, it's [stop it len...slap...thanks, i needed that]
    
    len.
    

    
    The proximity effect is more pronounced with dynamic mikes, like
    the SM series. It takes a certain amount of energy to get that 
    transducer moving, the biggest amount just to get it started. As
    you move closer to the mike, the energy level at the transducer
    increases, hence it gets louder. Low frequencies mean longer
    excursions, which take more energy. If the transducer is already
    moving, then it takes a proportionally smaller amount of energy
    to make the longer low frequency excursions than to start it from
    rest.
    
    Spatial relationships? Like wow man, pass the windowpane 8^)
    
    CdH
    

    
    My advice to .0 is to get a compressor, like the dbx 163x, it's
    only around $99.
    
    Paul Kent? mentioned an Alesis micro rack unit...sounds like just
    what I'm looking for. I was planning on a single rack space unit
    that held a Microverb and a 163x, but if Alesis makes one that is
    the same size as the Microverb, then maybe it'll be easier to 
    mount. Got any details on that little bugger?
    
    Thanks,
    
    CdH
    

    But!!!
    
    Given an environment with few or no reflective surfaces, it is
    still possible to tell which of 2 sound sources is further away,
    even if the volume of each is adjusted so that they appear to be
    at the same level.
    
    So what's the aural quality that allows us to make the distinction?
    
    Edd
    
    

Different microphone designs will have different frequency response curves at
different distances (due to proximity effect) and at different angles from the
centerline (due to the pickup orientation and porting).  With your microphone
you should have received some polar plots of its response curves.  A brochure
I have from one mic company gives these plots for each model.  I'll try
to remember to bring it in and type in some typical numbers.

    Re: mike proximity effect
    
    Most omnidirectional microphones have much less proximity effect
    than cardioids, if the excessive bass is a problem.
    
    I just stay a foot away when I sing, I don't have level problems,
    but then I'm not Aretha Franklin  8-)  8-)
    
    Re:  Alesis
    
    The Micro-whatever comp/limit box is $125 *list*, so is the exciter
    BTW and could be a reasonable alternative to the dBx 163.
    
    Re: ultra cheap peak limiter
    
    Find an old (like early 70s) cassette deck that has a switchable
    limiter...most old Sonys did, also a lot of other brands.  These
    limiters were designed to prevent overmodulation of the tape due
    to the fact that modern type II and type IV tapes weren't around
    then.  You can plug your mike into the deck, then run the deck's
    line out to your ministudio.....if you have a real board, plug the
    deck between the board and your multitrack.  Makes a very good stereo
    limiter.  Find a deck with bad heads, broken motors, etc. for $10-$25
    bucks and you're in business.

    
    As we all know sound travels in waves.  And since air is not a perfect
    medium, those waves disperse over distance/time due to friction.
    The effects the nodes and the cycles (the node is the height of the
    wave and cycle refers to the distance).  Because of the friction,
    the time to travel one complete cycle increases (distance/time/time)
    and the waves broaden (width).  Because of this effect more distant
    sounds will be naturally more resonant and carry less force.  Hence
    less impact against the microphone.  This will change the
    characteristics of the sound enough for the human ear to relate
    the distance.  And because the primary function of audio equipment
    is to repeat these waves, we can tell by listening to recordings
    which sounds are more distant.
    
    One simplistic example is to listen to an above ground train coming
    in an open area.  Even though the ground can be considered a reflective
    surface, you can still notice the effect described above.  This
    is also why live recordings made of a band at a concert take on
    a different quality (? correct term) than studio recordings.  In
    a studio recording all resonance is faked by reverb electronics
    and engineered reflective surfaces.
    
    As a note, the Rolling Stones recorded one of their albums (that
    I know of) in an old barn in NW Massachusetts, in order to get the
    more natural effect of spacial resonance.
    
    
    I know this is more physics than music, but what is music if not
    the physics of manipulating waves to create soothing sounds.
    
    
    David_in_C_Major
    
    

    re .23 - I think you mean "reverberant" rather than "resonant".
    Resonant means characterized by some kind of frequency peak.
    The reverberance is due to sound waves from the source taking
    alternative paths to the destination, and the effect is more pronounced
    as the distance increases.
    
    len.
    

    more re .23 - What you're syaing is that the pitch of sounds decreases
    as they travel through air.  This is a characteristic of some
    transmission media, but I believe that acoustic (pressure) waves
    in air only decrease in amplitude as a result of dissipative effects,
    and don't exhibit the sort of wavelength broadening (frequency
    dispersion) you've described.
    
    len.
    

    Len,
    
    Aha, you mispelled... well you... I mean...
    
    Ah, nevermind  ;-)
    
    	db

    Dispersion in transmission media is the tendency of different
    frequency signals to go different speeds.
    Tom

    
    Re: whoever it was
    
    Thanks for the Alesis tip. I'll call a few places and let folks
    know the details.
    
    Re: Engineer Tom
    
    What you say is true, although the speed isn't as important as time
    (which is, of course, a function of speed).
    
    Re: Edd
    
    2 speakers, one in front of you, one placed back a ways, levels
    adjusted to the same output. You are listening to the closer one
    in the nearfield, which is primarily the speaker. With the other
    one, you are in the reverberant field, which is some combination
    of the speaker and the room, depending on the throw characteristics
    of the speaker, the size of the room, where you are in it etc. The
    room has reflective and absorptive surfaces, which will alter the
    frequency response and phase of the signal. The ear is extremely
    sensitive to small changes in both.
    
    I like notes that have three conversations going on at once 8^)
    
    CdH
    

    
    
    RE .25
    
    Len,
    
    	I may be wrong on the use of RESONANT, I'll go back to my books
    and check, but if I remember correctly, pitch will decrease over
    time/distance, I'll check that also.  You have to realize that I
    took my physics courses waaaaaayyyyy back in college, and I am very
    rusty.  I was surprised at how much I did retain after this long
    considering the amount of permanent retention one can expect from
    college.
    
    	One more thing (since you seem to know something about this),
    how do you explain Dr. Bose's acoustic wave theories???????????
    
    
    They are supposed to work on these principles.
    
    
    David_in_Cb_after_time_and_distance
    
    

The following information is taken from an Audio-Technica pamphlet on their
"Artist Series" microphones.

Their various vocal microphones show elevated response centered on 150 to
200 Hz when you get closer than 12 inches.  At 3 inches it goes up by 5
to 7 dB, and at .25 inches it goes up by around 10 dB.

Some of the vocal microphones have a rolloff below 150 Hz, dropping by 15dB
at 50 Hz.  They also show a 6dB peak around 4kHz.  They have both dynamic
and condenser units with the low-end rolloff and high-end peak, although
they also have some condenser units that are essentially flat (except for
the low-end proximity effect).

They also have a line of instrument microphones.  The instrument mics typically
do not have wind screens, do not have bodies designed to minimize handling
noise, and have much flatter response curves (except for proximity effect).
In fact some of the instrument microphones show greater proximity effect
than the vocal mics; like +15dB at 100 Hz at a distance of .25 inches. 
Greater than 12 inches they are all pretty flat.

All the directional microphones have polar patterns that show that they are
most directional at high frequencies (5-8 kHz) and much less directional at low
frequencies (100Hz - 1kHz), with from-the-back response varying by 10-20 dB
depending on the frequency. 

For example, the ATM21 cardiod pattern dynamic instrument microphone rejects
rear sounds by 10dB at 100 Hz, by 15dB at 1kHz, and by over 20dB above 5kHz.

    re some things back there somewhere.
    
    I don't know what Dr. Bose's acoustic wave theories are, so I can't
    comment on them.  I do know who Dr. Bose is, as I was an undergraduate
    at MIT when he taught EE there.
    
    I am surprised to learn that pressure waves in air change their
    frequency (lower).  I have never heard of this effect (in air -
    it occurs with many other kinds of waves, in other media).
    
    Tom's definition of frequency dispersion is correct, and since I
    knew that, I'm not sure why I used the term the way I did.
    
    Regarding db's observation, I suppose one might have misread "syaing"
    as a typo or spelling error, but I thought everyone understood that
    "sya" was an acronym for "say, yea assert", i.e., say strongly.
    
    len.
    
    

    Ah, now I know why I mentioned frequency dispersion - it results
    in pulse broadening, which, if you think pictorially as I am
    occasionally wont to do, looks like a longer wavelength, yea verily,
    a lower frequency.         
    
    len.
    

>        I don't know what Dr. Bose's acoustic wave theories are, so I can't
>    comment on them.  I do know who Dr. Bose is, as I was an undergraduate
>    at MIT when he taught EE there.

    He still does teach acoustics in the EE department!
    
    	- Rick (MIT '82)

    My brother the physicist (MS MIT 19something) says YAG lasers can
    change the frequency of the input; air doesn't change the frequency
    of the input; air does act like a filter; the proximity effect is
    probably the result of the direcionality of higher frequencies,
    which I think was mentioned in an earlier note.
    
    So, the way I interpret this, high frequencies are somewhat directional
    out of the mouth, so the microphone intercepts about the same
    amount of signal with small changes in distance near the mike.
    However, lower frequencies spread out very faster in all directions,
    so 1/r**2 applies to them.  1/r**2 doesn't apply as well to
    high frequencies near the mike because the mouth doesn't look
    like a point source that near the mike.  It's kind of a disk source.
    Right next to the mike it might look almost like a plane, and plane
    sources don't decrease intensity with distance at all.
    Hm.  That would boost highs too.  Well, it does, you're louder next
    to the mike.  
    
    Let's get a pressure transducer and map the power output of a singer
    around them.
Tom

    I never realized physics and rock and roll had so much in common...
    There IS more than meets the eye!!!
    
    Well guys my patience ran out, I picked up the Alesis
    compressor/limiter at Sam Ash/Hempstead for $119.00.  I haven't
    had a chance to play with it yet but the instructions read well...
    
    Thanks for the input and physics lessons...
    
    Ben.

"Proximity effect" is not so complicated as people have been guessing.

First of all, it only happens on directional microphones.  Omni-directional
mics don't have it.

Uni-directional (especially "cardiod") microphones have these little holes in
the back called "ports" that allow sound waves to reach the back side of the
element so as to cancel out any sounds coming from the rear.  The lengths and
diameters of the ports are such that low frequencies are reinforced.

    
    Re: "Eating the mic"
    
    This is done in live situations only, ussually with a dynamic
    type microphone.  If you don't, you can get alot of bleed through
    from the other instuments and the monitor mix on stage which will 
    cause feedback.  This way the sound engineer can EQ and set levels
    for vocal peak levels and minimize feedback.  Dynamic mics use
    a plastic diaphram that vibrates and are usually lower cost ($100) like
    the SM-57, ATM-41, SM-58, etc. 
    
    In recording, it's best to use a condensor microphone.  Its diaphram
    vibrates in a magnetic field.  I use an AKG-414.  It needs 48 volt
    'phantom power' which my mixer provides.  Vocals are done with the
    vocalist standing 1 foot away and wearing head phones to prevent 
    bleed through.  I always use a compressor to even out the peaks, like 
    a high sustained note, shouts, etc.
    
    It is true that the farther you stand away from the mic, the thinner
    the sound (less bottom end).  The AKG-414 is extremely sensitive.
    From a foot away, I can hear lips smacking, breathing, clothes
    rustling, and eye blinking (well almost).
    
    If you have needle type meters, they are are a little slow responding
    to peaks.  The needle doesn't have a chance to make it to the peek before
    the voltage drops, especially on percussive sounds.	
    
    Dave

    re .37 - The diaphragm in a condenser mic vibrates in an electric
    field.  That's why they call them "condenser" mics and why they
    need power supplies; once upon a time (showing my age again),
    capacitors (which don't rely on magnetic fields) were called condensers.
    
    The diaphragm in a dynamic mic is not likely to be made of plastic,
    which usually is a nonmagnetic material.  Dynamic mics rely on magnetic
    fields.
    
    The chief characteristic of condenser mics is that they're expensive.
    They generally exhibit better preformance than dynamics, but that's
    not to say that dynamics are junk.  There are some pretty good dynamic
    mics available.
    
    len.
    

>    < Note 1158.38 by DRUMS::FEHSKENS >
>                                   -< Nits >->>
>
>    re .37 - The diaphragm in a condenser mic vibrates in an electric
>    field.  That's why they call them "condenser" mics and why they
>    need power supplies; once upon a time (showing my age again),
>    capacitors (which don't rely on magnetic fields) were called condensers.
>    
>    The diaphragm in a dynamic mic is not likely to be made of plastic,
>    which usually is a nonmagnetic material.  Dynamic mics rely on magnetic
>    fields.
>    
uh, the diaphrams of all the dynamics mics I've disassembled (cheap
    ones) look like transparent plastic; they were attached to coil
    electro-magnets, which moved back and forth around a permanent magnet
    inside them.  The coil moving thru the magnetic field gets an induced
    electric current in it.
tom

    re .39 - yeah, ok, the point is something magnetic or conductive
    has to move in response to the sound pressure; Dave's remarks made
    it sound like it was the plastic that was inducing a current.
    
    Not that that's impossible; conductive plastics do exist, and promise
    to be more common.  See the latest Scientific American cover story.
             
    len.
    

>    < Note 1158.40 by DRUMS::FEHSKENS >
>                       -< OK, Diaphragm .NE. Transducer >-
>    Not that that's impossible; conductive plastics do exist, and promise
>    to be more common.  See the latest Scientific American cover story.
>             
>    len.
See PAiAs probably first VCO keyboard, conductive plastic under the keys.
    Where the key contacted the plastic chose a length of conductive
    plastic, therefore different resistance.
    Condenser mics are probably made from plastic plates, I'm guessing.    
    T.I. calculators use conductive plastics under the keyboard to
    contact the pcb matrix, and give a springy feel to the keyboard.
    I've taken them apart.
    Our lab coats are some % conductive fiber, for electro-static discharge
    control
    Tom

Regarding compressors to handle vocal peaks such as shouts, etc:

If you watch an experienced radio announcer at work, you will see him turn
or move away from the microphone when he speaks louder, and move in closer
when he speaks softer.

I remember once watching Garrison Keillor doing "A Prairie Home Companion"
(I was in the third row), and he did a bit which required a loud exclamation
of "Hey!" at certain points.  Just before the "hey" he would turn his head
so that he was shouting 90 degrees away from the microphone.

    
    Ribbon mikes are considered dynamic mikes, I'm pretty sure of that.
    The better ribbons make wonderful vocal mikes, they're smooth and
    very transparent, much like a good condenser. I use a Beyer M260
    and love it. Their main fault is susceptibility to overload, they
    won't take the heat miking a kickdrum, for example.
    
    CdH
    
    
    

    Once you've recorded vocals (dry, using a compressor so no level
    problems) how do you go about mixing the vocals with the music ?

    I want to use a bit of reverb on the vocals (using a Midiverb ), and
    EQ is available.

    In my case the vocals are on a track on the 244, feeding into a
    channel on a mixer with sends to a Midiverb and/or delay. Recieves
    are stereo from the midiverb, mono from the delay, and all into
    seperate receive channels.

    What I can't achieve is a "tight" vocal that sits in the center of the
    mix (which is where I want it). 

    - How would you pan the (stereo)returns from the reverb in the case
      of vocals ? 

    - Are there any obvious rules about what frequencies should be
      boosted/cut ?


    Thanks
    Zayed

    I'm anxious to hear what some of the experts in recording engineering
    have to say about this.  I didn't get the "home recording book" that
    I asked for for X-mas but here's what I know that might help you.
    
    One technique I'm familiar with to "bring out" the vocals is generally
    referred to as "ducking".
    
    The general idea of ducking is to compress the "background" under
    the vocals by using the vocal track as the controlling signal
    (the one that determines the amount of compression) and using
    the "background" as the signal that gets compressed.
    
    To do this, your compressor should have two features:  Obviously,
    it should have two inputs, one for the control signal, one for
    the signal to be compressed).
    
    If you're mixing in stereo (and I assume you have a two-chanel
    compressor), the compressor should also have a "link"
    mode which basically means the equal compression used for both
    channels (left and right).
    
    	db - whose singing is so bad he ducks his vocals UNDER
    	     the background rathe than OVER it.

    
    Re: .44
    
    What is not "tight" about your vocals?
    
    If you mean that the reverb/delay being panned hard to the left and
    right makes the voice sound "wide", then pan them towards the center
    (is this too obvious?).  Really extreme stereo pans usually sound
    horrible anyway. Having the reverb returns slightly off center will
    give a sense of body to the vocals without smearing them across the
    stereo image. 
    
    EQ?  I would EQ the basic track to cut out everything below 100
    Hz, roll off the octave from 100-200 Hz to taste, boost to taste
    in the 1K-2K octave (too much and it gets nasal), roll off the extreme
    highs (10K and above) and play with boost and cut from 2K-10K to
    taste (try to balance "crispness" against sibilance).  

    

    Another method of bringing out the vocals is to use a harmonizer
    in stereo.  That is, pan the dry vocals all the way to one side,
    and pan a slightly detuned (harmonized) vocal all the way to
    the other side.
    
    I personally don't like flanged or chorused lead vocals.  Although
    I am unable to explain why, radical flanging or choruses seem
    to cause the middle end to drop out very significantly and EQ'ing
    does NOT bring them back.  A very subtle delay effect works ok
    but doesn't contribute anything dramatic.
    
    Other techniques I aware of:
    
    o Octave doubling (use subtlely)
    
    o Mucking with the tape speed during recording
    
    o Obviously, double-tracking (it's hard to do if you don't want it
      to be noticeable).  If you can't do it, you  might try using
      a delay with very little modulation.
    
    o EQ-slicing (splitting up the audio band across channels)
    
    o Layering your voice with a synth pad.  I've experimented doing
      this to enhance my bands back-up vocals.  I basically "double"
      the backup vocals with a 100-voice chorus turned way down so
      that it is not audible as a "different timbre".
    
      I haven't experimented much doing this with lead vocals but I
      know it's being done a lot.  I think pretty soon, it will be
      fairly common to double a vocal part by using a pitch-to-MIDI
      converter.
    
    I consider these things to be "enhancing" effects in that it's
    supposed to make a voice sound better, but not appear to be an
    "effect" as would a slapback, or reverb, etc.
    
    	db

    re .47:
    
    Two notable historical applications of double tracking, manual and
    "automatic":
    
    The Beach Boys often double tracked their lead vocals; the consistency
    of (Brian Wilson's?) voice and style made the effect quite subtle.
    
    Del Shannon frequently "ADT"ed (i.e., automatically double tracked)
    his vocals by using a delay at around 50 msec.
    
    len.

    < Note 1158.44 by REVERB::HANNA "Why not ?" >
>    In my case the vocals are on a track on the 244, feeding into a
>    channel on a mixer with sends to a Midiverb and/or delay. Recieves
>    are stereo from the midiverb, mono from the delay, and all into
>    seperate receive channels.
>    What I can't achieve is a "tight" vocal that sits in the center of the
>    mix (which is where I want it). 

    Assuming you pan the main vocal channel straight up, the center
    is the only place it can be.  What may be smearing your track around
    the stereo image is that the stereo effects returns from your MIDIverb
    are probably preset to pan hard left and right, giving a nice wide
    reverb field.. if you plug the reverb outputs into actual mixer
    channels (rather than the effects returns) you can then use THOSE
    channels' pan pots to bring the reverb closer to the center.
    
    karl

    According to one of my books on recording, boost 5000 thru 7000 Hz with
    a good graphics equalizer. Oddly enough, this happens to be the point
    that products such as the Aphex Aural Exciter start thier harmonic
    boost at. Strange coincidence?? I don't think so. You might want to
    wander down to your local music shop that sells these & try one out on
    one of thier tape decks. I recently built one of these types of
    circuits (look somewhere in the vicinity of note 1773 for more info on
    these devices) & I'm amazed at what it does for the vocal track. It
    does it better than boosting 5000 thru 7000 hz, but if you got an EQ
    available, that would be where I would start (Boost it between the
    microphone & the tape deck, onto the actual recording, not
    afterwards!!). Watch out for the Exciters, however, you'll be surprized
    to learn that you can't use them more than once on somethings (vocals
    can use it twice, if you are careful), for example drum mixes that have
    cymbals in them (whoa does it work bad!!!).

								Jens

    Thanks to all for the replies. Its giving me a few things to go back
    and try.

    I think the problem is with the way I'm panning the returns from the
    midiverb (hard left/right). I've done that mainly to give a nice spacy
    feel to the synth pad which does work fine. 

    But I find I lose the nice Stereo prespective when I mix down to
    two-track cassette if I *don't* have the reverb panned at least
    9 o'clock/3 o'clock especially for synths and guitars. Looks like
    I may start looking for a 2nd reverb to solve this.

    Zayed

    
    Try putting the reverb onto the vocal when you record it (mono)
    then add only a touch of reverb during mixing.  This will give a
    little bit of stereo spread to the vocal but still plenty of reverb
    on it.
    
    I assume you're using a four track??   I record all my tracks with
    effects because of the very problem you have.....the reverb or delay
    needed for each part is usually different.  It's riskier, if you
    have too much effect on the track you can't trim it back in the
    mix, but after you've been doing it for awhile, you get a good feel
    for what the tracks will sound like in the mix.

    I second that motion, I always put the effects on dyuring hte recording
    process to eliminate the need to attempt to get a reverb patch that
    is essentially a 'one size fits all' effect.
    
    Another thing I do since I use what is essentially a live vocal
    mike to record is take the bass nearly all the way off, and fuss
    about with my parametric midrange and highs to get a clearer sound.
    Live mikes tend to be muddy and colored and difficult to record
    with.
    
    dbii
    

    
    
    The other thing that  that should not go unmentioned
    is the *subtle* use of effects. There are very few voices that sounded
    good with the same sound of reverb as a synth pad. A lot of reverb
    is bound to make a voice sound muddy. I therfore half agree with
    the other comments on adding reverb to tracks on record. What I
    tend to do however is record all my instrument tracks with effects
    and then record the vocals dry. This allows me to experiment a little
    with reverb placement and presence during the mix on that important
    vocal track. I wholly concur with the comments on the exciter. Use
    it on record with the compressor and then you can just worry about
    a bit of gliss at mix time. 
    
                                Paul.
    				

    If I had a large number of compressors and channels, I would tend
    to moderately duck the reverb under certain kinds of tracks like
    vocals and guitars.
    
    In fact, if I was in a position to "lobby" reverb makers into providing
    new features, I would try and convince them to provide programs that
    do that automatically.
    
    When I'm not using my compressor and (soon-to-be-bought) extra reverb
    unit for recording, they will be in my guitar efx rack dedicated
    to my "ducked reverb" effect.
    
    	db