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

Bryan,

I do use 24-bit recording, and from what I've read, the RME Fireface's preamps 
and converters are pretty high end. Yes, no? Having said that, I notice that I 
need to turn the gain input on the fireface to about 75 percent in order to get 
this female's voice to reach about -7 on the peak meters in sonar. Any thoughts?


Omar Binno

Website: www.omarbinno.com
AIM: LOD1116
Skype: obinno1
  ----- Original Message ----- 
  From: Bryan Smart 
  To: ddots-l@xxxxxxxxxxxxx 
  Sent: Friday, July 24, 2009 5:22 PM
  Subject: [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
  AIM: LOD1116
  Skype: obinno1


------------------------------------------------------------------------------
  Windows LiveT Hotmail®: Search, add, and share the web's latest sports 
videos. Check it out.

Other related posts: