dbfs levels for rhythm guitar

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cliffdove

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What kind of levels do the pro audio engineers shoot for when tracking a mono distorted rhythm guitar track? So much so that it allows headroom left for drums, bass and the compressor in the master bus.

Most interface manufacturers suggest tracking the hottest possible signal at 0db, but from what I read, that's way too hot and should go for -25 dbfs to -18dbfs instead. Is there a minimum db level for metal guitars to sound really good?
 
I'm for recording the hottest signal at 0 dBFS, then you can always set your volume in the mix with the fader, right? I'm not sure if I understood well what you were asking...
that said, I'm absolutely no pro, and not even close to have made a really decent recording
 
as already mentioned, i also try to capture the signal with 0db level - of course the signal will vary (headroom)

when setting the mix with the other instruments i harmonise the eq with the key of the song, some threads down there is also an eq topic and link, very useful for your question

panning the rhythm git a bit out of centre can be charming as well. anyway it depends on the intention you follow with the sound.

stay tuned!
 
The AVERAGE level coming into your sequencer should be around -18dbfs with the fader at unity gain. This will include some peaks at around -16dbfs or so. Some mastering engineers prefer it in the -20dbfs or -22dbfs range.

Anything hotter than this and you're likely to encounter the dreaded digital clipping.

I'm surprised that this basic concept is left out of manuals for sequencers.
 
Rotary is on this one...
Equipment has a nominal level where it works best. For most, it is now 0bdvu which equates roughly to -18dbfs on DAWS. You will hear the argument about not using full bit depth if you don't record closer to 0bdfs. It may have been more of a problem with 16 bit recording, but it isn't with 24 bit. If you're dealing with an analogue input, you are still dealing with analogue acoustics. Digital will record perfectly right up to clipping, then it is a hard clip. Analogue won't fail as quickly. It will be something more like: perfect, less focused, grainy, distorted, hard clip. I think the analogue to digital conversion process is what confuses people. There is a great thread on Tweak's forum:
http://studio-central.com/phpbb/viewtopic.php?t=34078
 
If you're tracking at 24 bit depth, there's no need to track at hot levels or
use analog compression that gets printed to disk. Just leave your track faders
set to 0db & turn down the preamp gain so the peaks don't exceed -6db.

Tracking at 16 bit depth is a different story; stay as far away from the noise
floor as you can without clipping; it offers less headroom than 24. A good range
for peaks in a 24 bit depth stereo mix before mastering is -10dBFS to -6dBFS,
while 16 bit should run at -6db to -3db.

Keep mixing levels between -10db to -6db to allow room for mastering.

If the multi-track mix doesn't sound loud enough to you at -10db, turn up your
monitor.

At mastering, the final level is raised to compete with commercial CDs with a
limiting threshold of -3db.

Many mastering engineers working with digital equipment would agree that a minimum of 3 to 6 dB of available headroom is critical to perform good mastering. Ideal peak levels should not exceed -3dBFSD & the average sum of the left & right channels should be at around -10 to -18 dBFSD.

Mastering

1. Transferring the recorded audio tracks into the Digital Audio Workstation (DAW) (optional).

2. Sequence the separate songs or tracks (The spaces in between) as it will appear on the final product (for example, an Audio CD).

3. Process or "sweeten" audio to maximize the sound quality for its particular medium.

4. Transfer the audio to the final master format (i.e., Red Book-compatible audio CD or a CD-ROM data, half-inch reel tape, PCM 1630 U-matic tape, etc.).

Actions taken during mastering

Edit minor flaws.
Apply noise reduction to eliminate hum and hiss.
Adjust stereo width.
Add ambience.
Equalize audio between tracks.
Adjust volumes.
Dynamic expansion.
Dynamic compression.
Peak limit the tracks.

The key is not to overcompress your mix prior to mastering. Preserve dynamic
range - which means don't turn down the volume unless you have to to avoid
clipping.

You can also use a compressor with a sidechain so that compression is triggered
on the instruments whenever the vocal comes up. When the vocal comes in, very
slightly duck all tracks but the vocal, which you ride up a notch or two. Subtle
adjustments are key here. Like when drums come up slightly during flashy fills,
the bass drops back when the string swells build up, the guitar riff leads off
right out in front, then gets pulled back on the slider & with a hint of
reverb as the it repeats through the prechorus, etc.

Allow for a touch more headroom than specified to allow for disparate hardware.

To insure adequate gain structure in your inputs, set track faders to unity &
adjust your input gain just below clipping or zero level (about -2db). After all tracks are
configured, set the main faders to unity & adjust the relative levels of each
track for the best mix. Finally, set the main faders to -6db output. Adjust
monitor volume to taste.

Close down all of the trim pots, faders, power amplifier, & record level
controls. Set the master fader to the 0 dB position (or the marked normal
position). Start playing a line level source attached to one input. Open the
channel fader for the source to the 0 dB or normal position. Adjust the trim for
an average level of -10 dB on the meters. Stop playing that source, & start
another. Repeat steps 4...6 for each source. Play all sources at once, & touch
up trims for initial balance & proper peak levels. Set the record level on the
recorder, if it is external to the mixer. Set level controls on PA & monitor
amps, if applicable.

Vocals should typically have ratios between 2:1 & 4:1.

Summing tracks to a bus adds +3db/track, theoretically.

Before beginning to mix, play back your tracks with all faders pull completely down. One-by-one push each fader to maximum (+10); adjust that tracks trim control so that the playback meter reads your desired setting (-1); pull down that track's fader & move to the next one. When you've completed every track, you can move all faders up to nominal level to hear all tracks played back with equal volume.

Note the trim setting on each one (they'll differ depending on how they were recorded & the sound of the instrument/voice). Trim adjusment allows you to minimize the fader distance between tracks for mixing...making it easier to control (ride) the mix. Also, at higher fader positions fader movement is more precise than coarser movements in the lower positions, i.e., at the high fader position you can make refined adjustments. Volume balancing is a different process, but good volume balancing begins with gain leveling via trim adjustment.

Understanding dB

dB is an abbreviation for "decibel". One decibel is one tenth of a Bel, named for Alexander Graham Bell. The measurement quoted in dB describes the ratio (10 log power difference, 20 log voltage difference, etc.) between the quantity of two levels, the level being measured & a reference. To describe an absolute value, the reference point must be known. There are a number of different reference points defined. Here are a few: dBV represents the level compared to 1 Volt RMS. 0dBV = 1V. There is no reference to impedance.
dBu represents the level compared to 0.775 Volts RMS with an unloaded, open circuit, source (u = unloaded).
dBm represents the power level compared to 1 mWatt. This is a level compared to 0.775 Volts RMS across a 600 Ohm load impedance. Note that this is a measurement of power, not a measurement of voltage.
dbFS - relative to digital full-scale.
dB SPL - A measure of sound pressure level.

A few easy-to-remember facts that may help: If you're dealing with voltage measurments, convert from dBV to dBu: 1dBV equals +2.2dBu.
+4dBu equals 1.23 Volts RMS.
The reference level of -10dBV is the equivalent to a level of -7.8dBu.
+4dBu & -10dBV systems have a level difference of 11.8 dB & not 14 dB. This is almost a voltage ratio of 4:1 (Don't forget the difference between dBu & dbV !!)

dBFS - dB Full Scale

0 dBFS represents the highest possible level in digital gear. All other measurements expressed in terms of dBFS will always be less than 0 dB (negative numbers).
0 dBFS indicates the digital number with all digits ="1", the highest possible sample.
The lowest possible sample is (for instance for 16 bit audio):
0000 0000 0000 0001, which equals -96 dBFS. Therefore the dynamic range for 16-bit systems is 96 dB. For 20-bit digital audio it is 120 dB. For 24 bit digital audio it is 144 dB.
Full-scale input level is the analog input voltage level that will cause the A/D converter to just equal full scale with no clipping on either positive or negative peaks.

Output full scale is defined as the analog output voltage produced while playing a 997 Hz digital full-scale sine wave, assuming the THD+N is less than -40 dB relative to the signal level.

The dynamic range of a digital system is the ratio of the full scale signal level to the RMS noise floor.

Studio Mix Levels Short Answer:
A studio mix should have peaks between -10.0dBFS & -3.0dBFS to allow headroom for mastering. A good mix with peaks at -6.0dBFS should put your RMS between -18dBFS & -12dBFS for most pop & rock music.
http://www.cakewalk.com/forum/tm.asp?m=81256&mpage=1&anchor#81888

Studio Mix Levels Long Answer:
http://digido.com/portal/pmodule_id=11/pmdmode=fullscreen/pageadder_page_id=119

Final Master Levels Short Answer:
Don' t flat top your masters with a brickwall limiter. Leave room for dynamics. Final mastering should have peaks limited to -0.3dBFS to allow for intersample peaks that may not be revealed by your application meters, or that may be exaggerated during any sample rate conversion (SRC) you may be doing for the target medium.
http://www.cakewalk.com/forum/tm.asp?m=81256&mpage=1&anchor#81850

Final Master Levels Long Answer:
http://www.digido.com/portal/pmodule_id=11/pmdmode=fullscreen/pageadder_page_id=59
 
DeFrag said:
If you're tracking at 24 bit depth, there's no need to track at hot levels or
use analog compression that gets printed to disk. Just leave your track faders
set to 0db & turn down the preamp gain so the peaks don't exceed -6db.

I have reason to believe due to the nature of the genre of music I'm playing, metal rhythm guitars simply have to be tracked hot. After all the tracking level tests that I've done at -22, -18, -12 and as hot and close to 0dbfs without clipping, my ears are telling me that 0dbfs sounds best. I do find that mixes that were tracked at -18 tend to be more 'jumpy' and dynamic because of the headroom available. Even if I apply compression and limiting in the master buss close to 0, it just doesn't sound as good with instruments that are tracked hotter to achieve overall punch and loudness.

I've also asked many sound engineer friends their take on this and was told that its best to track as hot as possilble because the quality of the audio is just shitty at -18.

Nonetheless, I'm willing to try many approaches and whatever it takes to understand the grand scheme of things. Will give it a shot and try your suggestion of peaks not exceeding -6.

Cheers,
 
Recording and levels are subjective at best. I have spent 100s of hours in world class studios and there is absolutely no rule of thumb which specifies or even suggests a specific level for recording a guitar or any other instrument.

Point 1; The trim knob on a channel strip is used to set the maximum signal strength coming into the channel. Bring the fader to 0db and trim the input signal to a point where you are just below clipping, supposedly 0db, but usually higher depending on the board used. If you don't know what that is it's the point where the indicator turns red or you hear the channel distort. Don't adjust the signal levels any further using the trim pot. Use the faders, that's what they're for. This is considered unity gain.

Point 2; Record without effects or compression. Compressing the input signal will always lead to a loss of tone and depth. Don't put something in that can't be taken out later. If you use effects or compression after the track is recorded and don't like it you can always remove it. Record each instrument to it's own track if possible. If not record drums and bass together, then keys, then guitars one at a time and finally vocals. Keep notes on settings.

Point 3; The microphones used and how they are placed have more to do with the actual recording than the signal strength. Recording hot (above 0db, moderate clipping) is a hacks method used to overcome the problems inherent in a poor quality microphone or poor placement and technique. Buy a decent mic for yourself. Start by trying a Heil PR-30 or PR-40, the same mics used to record the sound for the movie "Letters from Iwo Jima". But you must, regardless of the mic used, experiment with the mics position. Here's a clue. The mike should almost never be facing the cabinet straight on (bass cabinets are an exception), because the middle of the speaker cone is usually a dead spot.

http://www.heilsound.com/pro/products/

Point 4; Sound is measured in decibles, db. Buy a cheap Radio Shack db meter and bring the volume of your guitar to about 100db when measured from 3 feet away. That's more than loud enough to get great sound and tone from any cabinet.

Point 5; Dump the budget headphones and get a pair that allows you to hear what is really being recorded. The AKG 240 series are a studio standard and pretty cheap at $100. Buy a pair.

http://www.sweetwater.com/c412--AKG--Headphones

Point 6; You are not limited by the equipment you use, you are limited by how you use it. Almost anything can make a good recording today and although 16 bit systems have limitations prisitine recordings can still be made with 16 bit equipment. Use the right mikes, know how to use the board, software or desk, and you're all set.

Have fun with these basic tips.
 

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