questions from old musician/newbie engineer

[willow]

i will be getting my phonic 18 ch. firewire mixer monday. i have a new puter (HP dualcore AMD) ..couldn't get a custom built for many reasons.

anyway, as i noted in previous posts, i have been playing drums for over 30 years, and have done alot of recording, but i always just sat down and played and never really took part in learning much about engineering, especially in the digital domain.

i have a big drum set, but i probably will use only snare mics (2 snare drums) and a bass mic and two overheads.

i have so many questions, but i will start with just a few at a time.

this thread may go on forever and i appreciate the help from you guys!

ok, lets start with panning. i know about mic placements on the drums, how the different drums can be placed in the stereo field and using overheads for left and right hard panning. but my question is... panning is better done at the recording phase? if i set all the pan parameters when i record, then when i'm mixing later in sonar, i would leave everything in center, right?

what about if something was recorded without the panning set (everything in center)and you adjust it later in sonar. what differences occur with this method as opposed to the above?

another thing i'm wondering about is... with two consoles (one real and one in sonar) how does that work? does the real one control the one in sonar?

did i phrase these questions right?

thanks

[FALKEN]

you are probably better off leaving everything up the center while tracking. this will help you to hear any phasing problems or any other problems more easily. the general rule is that if it sounds good in mono, it will sound better once you pan it.

as far as the firewire mixer, i've never used one, but I imagine you can use it as a control surface in sonar? you're going to have to read the manual on that one. maybe someone with experience with your brand of mixer can help. perhaps the manufacturer has a forum as well?

[willow]

my next two subjects are phase and meters.

phase...

recording - how do you avoid phasing (mic placements)and when do you mess with the phase inversion option? are there certain settings that are better before recording?

playback- how do you know when to set a certain track with inversion? for example.. i have a song that was recorded with 8 mics. an engineer heard this song a while back and he said that there were phasing problems in the drum tracks. how did he go about distinguishing them and how would you correct it? is it a matter of finding a track that sounds weak and just inverting that one track?

meters- i remember recording on reel-to-reel and i was pretty good at getting good input levels just by hitting in the red only on quick peaks

recording- in sonar, i usually let the meters go to the red occasionally on the fast peaks. (not all the way... -6 area) should i be recording a little hotter?

playback- when i play all the tracks.. the master is too loud when it is set on 0 db? would it be best to set all the tracks at levels so that the master is showing optimum levels at 0 db?

i hope you understand the questions the way i'm asking them..

any info and "rules of thumb" would be appreciated

gracias

[ez_willis]

Check out this link for some great info on mic placement.

Regarding digital recording levels, you need to keep away from those peaks(red). Digital distortion will make you want the reel to reel back.


Check out Mixing 101 from Blue Bear.

Use the search function on the board as well

[chessrock]

Jesus, dude.

You want someone to answer all of that in one post?



Why don't you just ask how to split an atom, and then re-fuse it?

.

[willow]

Originally Posted by ez_willis

Check out this link for some great info on mic placement.

Regarding digital recording levels, you need to keep away from those peaks(red). Digital distortion will make you want the reel to reel back.


Check out Mixing 101 from Blue Bear.

Use the search function on the board as well

thanks for the info,

so, you are saying ... NO red at all? not even quick jumps?

[ez_willis]

Originally Posted by willow

thanks for the info,

so, you are saying ... NO red at all? not even quick jumps?

Anything at or above 0dB in the digital realm will result in the audio equivalent of dentistry without anaesthesia.

Audio volume is measures in decibels, dB. With analog audio it was determined and standards were set so that a reading of 0 dB on a VU meter was the highest volume level before audio distortion sets in. Audio that exceeds 0 dB is called 'overdriven' audio. It's at 0 dB that analog audio starts to distort. The higher the dB reading the more distortion. But analog audio is very forgiving. Going +2 or +3 dBs over 0 results in very minimal distortion, often described as warm or fuzzy.

Not so with digital audio. Digital audio is totally unforgiving in this respect. Once your volume level hits 0 you experience what is called 'digital clipping'. You must keep the very loudest part of your audio (spikes) below 0 dBs.

[willow]

yes, i do the "search" thing on all of my questions, but it seems that i never find exact answers to my particular questions, or info is usually written with the assumption that people already know certain things. that is also why many manuals are hard for people to understand.

ez... thanks for helping me with this meter thing. i'm still not completely sure about this, so let me break it down..

now, i already knew that no signals should reach 0 db

but, about the meter in sonar ... the red starts about -6

ok, now speaking of the "red zone" (-6 to 0db) it is ok for occasional spikes to go in the red (not to the top, just around -6) right??

[slidey]

as long as it stays below 0db.................I try for happy medium sake try & keep the levels at around -10db-6db so if a majorly loud hit comes in it won't peak over 0db

as pointed out earlier digital distortion sucks in a big way

[ez_willis]

Originally Posted by willow

yes, i do the "search" thing on all of my questions, but it seems that i never find exact answers to my particular questions, or info is usually written with the assumption that people already know certain things. that is also why many manuals are hard for people to understand.

ez... thanks for helping me with this meter thing. i'm still not completely sure about this, so let me break it down..

now, i already knew that no signals should reach 0 db

but, about the meter in sonar ... the red starts about -6

ok, now speaking of the "red zone" (-6 to 0db) it is ok for occasional spikes to go in the red (not to the top, just around -6) right??

I'm with ya now.

I'm not familiar with Sonar, but it sounds like "the red zone" is approaching digital clipping, not actually over 0dB. As long as you don't hear a sound similar to GGGGRRRRRANNNZZZZZZNNNNNNNNNNSSSSTTT you're okay.

[SouthSIDE Glen]

Originally Posted by willow

recording - how do you avoid phasing (mic placements)and when do you mess with the phase inversion option? are there certain settings that are better before recording?

This is a very complicated question which can only be completly answered with an answer the size of a paperback book. But here's some short answers to get you started:

One should typically only be *planning* to phase invert a mic signal when that mic is tracking something where the physical waveforms themselves are inverted from the signal on another mic. One typical example would be when one is miking both the top and bottom of a drum. When the stick hits the drum, the top skin is pushed down away from the top mic, but the bottom skin (or the air inside the shell if there is no bottom skin) is pushed down toward the bottom mic. This is a physical world version of a phase inversion. therefore one would invert the signal on the bottom mic, essentially inverting the inversion and synching the waveform back up with that of the top mic.

The commonplace general rule to minimizing other phase issues is what's usually referred to as the 3:1 rule. This basically says that when you have two mics miking two sources (e.g. two drums, two horns, one guitar and one piano, etc.) that the distance between the two mics should be at least three times farther than the distance from mics to the instruments they are miking.

Originally Posted by willow

playback- how do you know when to set a certain track with inversion? for example.. i have a song that was recorded with 8 mics. an engineer heard this song a while back and he said that there were phasing problems in the drum tracks. how did he go about distinguishing them and how would you correct it? is it a matter of finding a track that sounds weak and just inverting that one track?

It's hard to put into words what phase problems can actually sound like, he just had the experienced ear. As far as correcting it, see the 3:1 rule above for starters.

Second, make sure you are using directional (cardioid) mics and set them up in such a way as to minimize the amount of bleed you get from other drums.

Third, cut down on the number of mics. I'm serious. There is no need or reason for a "newbie engineer" such as yourself to be jumping into engineering right off the bat with both feet into the deep end by multi-miking the drum kit. Honestly, I don't know where you guys all seem to get the idea that it's necessary to use so many darn mics on a single drum kit. It only takes two ears for a drum kit to sound good live, why should it take 8 mics to get it to sound good on a recording? Answer: It shouldn't and it doesnt. If after you've been at it for a while and have some miking technique experience going for you, you are welcome to expiriment with multi-miking technique. But in the meantime, start with a couple of overheads or a front stereo pair and add a kick mic as the third track. Some of the best commercial albums and CDs have been recorded this way. Mabe then you can add a single snare mic if you want, and then move on from there, but much of the time even that is not necessary.

Originally Posted by willow

meters- i remember recording on reel-to-reel and i was pretty good at getting good input levels just by hitting in the red only on quick peaks

recording- in sonar, i usually let the meters go to the red occasionally on the fast peaks. (not all the way... -6 area) should i be recording a little hotter?

playback- when i play all the tracks.. the master is too loud when it is set on 0 db? would it be best to set all the tracks at levels so that the master is showing optimum levels at 0 db?

The term "dB" or "decibel" alone is not enough these days, one has to define on what scale they are being measured. It's like saying "32 degrees". Depending on whether your talking Kelvin, Farenheight, or Celsus, 32 degrees can mean deathly and unearthly cold, or the temperature of ice, or hot enough to run naked on a sunny beach. It's the same with dB.

In the analog realm the meters are typically referring to dBVU, in which case 0dBVU is calibrated simply to match the level at which the voltage matches the rated input or outpt level voltage specification for that device (or for tape recording, 0dBVU is usually calibrated during recording to match the rated magnetic saturation level for the reference tape recommended for that machine, as measured in nanoWebers/meter.) The reason it's usually safe to push an analog signal a bit past 0dBVU is because there is still some "headroom" left in the physical tolerances of the circuitry (or tape head or tape) beyond the rated/calibrated specification. Tape oversaturation or the overdriving of tube voltages cause the "analog distortion" that some find "pleasing"

in the digital realm, it's a different story, however. There things are usually measured in dBFS. "FS" means "Full Signal" or "Full Strength". On this scale there literally is no such thing as +1dBFS. 0dBFS is as high as things will go. Period. This is because 0dBFS represents a digital signal of all "1"s, and there is no way to go higher than that digitally. You push a digital signal harder than that and you simply run out of "1"s. There is no more "headroom" like there is in the analog domain. What happens is anything that tries to go higher than that just gets clipped off at 0dBFS. This is clipping distortion. Can one sometimes momentarily clip? Yes; small amounts of digital clipping distortion can be gotten away with without becoming audible to 99.999% or the population. But too much of it and it becomes extremely audible, and not in a good way by any measure. But though small amounts can be gotten away with, that is sloppy technique; there is really not much of a valid reason to do so. One can acheive sumilar volumes and results without that clipping if they learn how to gain stage, mix and master their stuff the right way.

G.