[ddots-l] Re: question about microphone gain input

Just wanted to point out that these sorts of situations are one of the
reasons why high-end preamps and 24-bit recording is superior to budget
pres with typical 16-bit recording. With a budget setup, even with a
quality mic, you're always fighting the noise floor (the background
hiss/hum/buzz coming from the analog part of your recording setup), and
dithering artifacts (the extremely quiet digital noise that is the
result of attempting to record signals that are more quiet than the
sampling resolution can accurately represent in digital form).
 
Here are a few interesting facts about levels and digital recording.
 
First, about levels. You probably know that the strength of a signal is
measured in DB, with 0 being the loudest possible signal that can be
accurately stored in digital form. As the number decreases, the level
drops off on a logarithmic, not a linear, curve. This is important to
know. If, for example, you lower a signal by 6 DB, the signal will sound
half as loud as before. For each additional 6 DB, the signal will be
half again as quiet as before. The same idea works the other way around:
if you raise a volume control in Sonar by 6 DB, the track will sound
twice as loud as before.
 
This fact about levels is directly related to how the signal is
digitally represented. As you might know, the computer represents all
sound data in binary (as a long pattern of 1s and 0s). If you are
recording in 16-bit, then the computer uses a group of 16 1s and 0s (or
16 bits) to describe each digital snapshot/sample that it records from
your audio interface's inputs. If you are recording in 24-bit, then the
computer uses a longer group of 24 1s and 0s to represent each sample
that it takes from the audio interface. Regardless of the length of the
group of 1s and 0s, the first few bits always describe the portion of
the sound that is the loudest, and additional bits describe parts of the
sound that are increasingly more quiet. Specifically, the second bit
describes the part of the sound that is half as loud as the first bit,
the third describes the part of the sound that is half as loud as the
second, and so on. Also, each additional bit has a more difficult time
at accurately representing its portion of the sound. Even cheap
converters will encode the first 12 or so bits accurately, but, beyond
this point, the bits are representing sound that is so quiet, that
occasionally what should have been a 0 is stored as a 1, and vice versa.
These little errors aren't usually a problem, since they're so quiet,
that you don't hear the errors in the sound....until you amplify the
sound, that is. Even if you have great mics, great preamps, and great
converters, there comes a point where there aren't enough bits available
in the sample to represent sounds that are quiet beyond a certain point.
 
So, when you reduce the level of the signal coming in by half, that is a
gain reduction amount of -6 DB, and that also means that the first bit
of the digital sample is no longer used. If you reduce the incoming
signal's level by another half, to a quarter of full strength, that is a
total gain reduction of -12 DB, and now the first two bits aren't used.
If you recorded a signal at this level with only 16-bit resolution, the
converters on your audio interface would only be using the quietest 14
of the 16 available bits that you're using to record the sound. It would
be very likely that a part of what you recorded at this level was so
quiet that it was beyond the lower limit of the converter to represent
in only 16 bits. you could digitally amplify/normalize the recording
back up to 0 DB, and the signal would sound loud, but it is still only
recorded with 14-bit resolution. This is where the digital artifacts
come in. When Sonar amplifies the signal, it basically has to promote
bit 3 to bit 1, bit 4 to bit 2, bit 5 to bit 3, etc; all of the bits get
shifted to the left/up. However, Sonar still only recorded 14 bits of
signal for each sample, so it no longer has the last two bits to
accurately represent the quietest detail of your recording. You'll hear
some small amount of crackle/fuzz because the bits that were less
accurately represented have been promoted to represent louder sound than
they originally recorded.
 
This can be bad if you have good converters, but, if you have cheap
converters, it can be horrible. Remember that I said that many cheap
converters don't accurately represent data beyond the first 12 bits or
so? Well, if you have one of these, and you normalize the quiet signal
as I described above, you aren't turning a 16-bit signal in to a 14-bit
signal, you're turning a 12-bit signal in to a 10-bit signal!
 
I hope that this starts to make it clear why 24-bit is better than
16-bit. If you recorded an extremely quiet signal, as we discussed
above, and had to boost it by 12 DB, then you'd still be giving up the
same two bits. However, if you'd recorded in 24-bit, instead of
recording in 16-bit, then you'd be reducing your recording with 24-bit
resolution down to 22-bit resolution. This means that a 24-bit signal
that is amplified by 12 DB still has dramatically high resolution (about
64 times as much, in fact) than the original unamplified 16-bit signal.
 
So, in summary, recording in 24-bit allows you to dramatically amplify
the recorded signal without introducing a lot of grainy digital
artifacts. When you record in 16-bit, amplification really isn't a good
option.
 
Now, armed with that info, back to recording technique.
 
In budget 16-bit world, you get your levels set for your vocalist by
trying to have them sing as loud as possible, and adjusting the gain on
the input channels so that the meters never go over about -3 or -4
during that absolute loudest note. As you've discovered, if you attempt
to boost the recorded signal after the fact, you increase the level of
the dithering noise. It is also tempting to close-mic the performer
since, if you don't, you must turn up the input channel's gain in order
to get the meters close to -3 or -4, and, the more you turn up the input
gain, the noise floor of the preamp also becomes louder. You can get
good results when recording at 16-bit resolution, but it takes a lot of
work to do it well. If you record too quietly, then you'll pay for it
with dithering artifacts when you amplify the recording during the mix.
If you record the signal as hot as possible, then you'll avoid dithering
artifacts, but you'll run a greater chance of clipping during a really
good take. At the same time, if you turn up your preamp to get a good
hot level, the noise floor of the preamp comes up and you hear hiss. If
you keep the preamp turned down, but close-mic the performer, you'll
have mic proximity effects to deal with, and, for vocalists, you'll risk
popped Ps, breathing in the mic, and other hazards that can ruin a take.
This is even more difficult when you're recording several
people/instruments at once, since you have lots of input levels that are
near maximum, and could clip at any time. You can use hardware limiters
on the inputs to prevent digital clipping, but they'll still color the
sound. Basically, you must really work and/or spend a lot of money on
gear in order to avoid problems at 16-bit.
 
However, suppose you had a high-end pre, a high-end interface with
quality analog/digital converters, and recorded at 24-bit resolution.
When you were setting up levels for your vocalists, you could let the
levels peek at -10 or lower. Instead of only 3 or 4 DB of headroom,
you'd have 10 DB to work with. This means that, in order to clip,
someone would need to sing or play a note that was over 2 and a half
times louder than the loudest note that they played when you were
setting up levels. This is highly unlikely. Once you've finished
recording, you'll still need to amplify/normalize the recordings, but,
even after boosting the signal by 10 DB, you still will be using a
recording with 22-bit resolution.
 
24-bit recording, while it will allow you to avoid clipping by recording
at lower levels and later amplify the recording with almost no
noticeable artifacts, won't save you from a noisy preamp. Preamps seem
to do an increasingly worse job with cleanly amplifying a signal the
more you turn them up. Besides having a nicer tone (which is a concept
that it is hard to describe), higher quality preamps can produce higher
gain for your converters without yielding a lot of hiss.
 
You're using a Tascam FW1884. The FW1884's built-in preamps aren't that
bad, but they put out a lot of hiss when you crank them up. You have
some options.
 
First would be to use better preamps. The FW1884 has an ADAT input that
will accept another 8 channels of digital signal from another piece of
digital equipment. You can get a dedicated high quality preamp/digitizer
unit and connect it to the FW1884 through the ADAT port. Presumably,
you'd buy a device that has preamps with greater dynamic range. Then,
you could crank up the mics without hearing lots of hiss from the
preamps.
 
Second, you should reconsider your mics. In the perfect world, you'd use
condenser mics to record your vocals. Good condensers will put out a hot
signal, and have a very low noise floor, so you won't have to crank up
the preamps very far, and, when you do, you'll have to crank them up a
long way before you hear any hiss coming from the mic. Unfortunately,
condensers are very sensitive, and so they're not as nice when you can't
isolate the performers. If you have a vocal booth/closet, then a
condenser mic might be something to think about. If you'll be recording
several people in the same room, then condensers won't work, since
they'll pick up everything in the room, not just what you point them at.
 
Still, even among dynamic mics, there are levels of dynamic range. If
you must use dynamic mics, then better preamps are probably the way to
go.
 
There is my essay for the week. *smile*
 
Bryan

 
________________________________

From: ddots-l-bounce@xxxxxxxxxxxxx [mailto:ddots-l-bounce@xxxxxxxxxxxxx]
On Behalf Of Stacy Blackwell
Sent: Friday, July 24, 2009 2:51 PM
To: ddots-l@xxxxxxxxxxxxx
Subject: [ddots-l] Re: question about microphone gain input


Omar, in addition to the other comments, I have just recorded 4 lead
singers in my band with the levels all supposedly set the same.  The
male vocals did not require much trim during playback, but the female
singer, well, I am having to really turn up the trim and now a high hiss
is heard when soloed.  I think she was scared of popping the mic and
held it too far away.  Also, if a person sings to the side or across the
top (depending on the type of mic) instead of directly into it, the gain
and tone can be affected.  Trial and error, live and learn.  But I do
think that singers, male or female, differ in their abilities to
"project" their voices which can result in different gains.  I have also
learned that I should also wear headphones to listen for a constant
entering volume from the singer instead of listening to it after the
recording.  Keep in mind that I don't have a "real" studio and have this
equipment mainly for my own personal songwriting and recording, so this
was my first time to record other vocalists.  See ya,  S.B.  
 

________________________________

From: omarbinno@xxxxxxxxx
To: ddots-l@xxxxxxxxxxxxx
Subject: [ddots-l] question about microphone gain input
Date: Thu, 23 Jul 2009 18:00:26 -0400


Hello,
 
I have my vocal mic plugged into my Fireface Soundcard. When one of my
female clients records vocals, I end up having to turn up the gain quite
a bit. However, I notice when I speak or sing into the mic, I need the
gain up only half way. Is it usually the case that gain input on the
soundcard or mixer changes from singer to singer, or does it usually
stay at the same level.
 
Thanks.
 
Omar Binno
 
Website: www.omarbinno.com <http://www.omarbinno.com/> 
AIM: LOD1116
Skype: obinno1

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