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Old 1st May 2007   #1
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Tracking - Levels

I've always been told, when tracking, that you want to get the "hottest" signal. So just make sure all signal/peaks are the biggest without clipping.

This is concerning rock/pop music.

Is this the correct way of thinking? Or do you want to give yourself some "headroom" on the channel?
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Old 1st May 2007   #2
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You'll get the best results when tracking levels are between -6 and -12 on the meters. Somewhere in between there with the peaks at -6 is perfect. If everything is tracked super hot you'll overload your master bus/fader and end up turning all the channels way down, which is turn causes bits to be lost in the digital realm (thus negating your tracking efforts).
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Old 1st May 2007   #3
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I try to keep the RMS around -18 to correspond with 0dBVU equaling about -18FS. Most digital converters (especially cheap stuff) will mangle a signal that gets even near it's max. Drums can PEAK around -10FS and I try to keep guitars/bass/vox around -18 to -12. When 16bits was big, getting a hot signal was more important. The 24bit world makes life easier.


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Old 2nd May 2007   #4
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Exclamation

Ok whoa... this puts a whole new perspective to things!

I have an Apogee Ensemble - and record everything at 48/24.

So you are saying I shouldn't have a signal at my coverter being any louder than -6db at the very most?

One of you are saying -6db at the most the other -10db (for drums) at the most?

Maybe this is my problem! Because I'm always having to push the faders down on each channel to avoid the master clipping.

I guess I'm living in the past... I thought you were always suppose to get it as loud as possible, thus tracking the best "tone".

This is something I would really like to discuss because apparently I have the wrong idea.

Thanks for the help guys.
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Old 2nd May 2007   #5
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It depends .. you certainly dont want to track so quiet that you have to turn your tracks up.
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Old 2nd May 2007   #6
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my peaks are at -3dB while tracking. if it is any sort of a problem, I like this problem (convertor nonlinearity.. )!

I lower levels while mixing (thanx terry manning). works very good!

cheers
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Old 2nd May 2007   #7
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Originally Posted by George Necola View Post
I lower levels while mixing (thanx terry manning). works very good!
What do you mean by this? do you link all the levels and turn them all down?

By the way... what is RMS?
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Old 2nd May 2007   #8
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RMS is root means square ==> kind of average of an audiosignal.

RMS describes the loudness (well.. this is not true, but my english is too bad to explain it in detail I think) of an audiosignal

I lower the faders in my DAW (the channelfaders) by 6dB.

because I route all the channels to groups, it's not a problem. I still get good levels at the groups, then i run not so hot into the masterbus and push the signal there.

cheers
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Old 2nd May 2007   #9
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Quote:
Originally Posted by RadioMoo View Post
Check out this thread
Oh Lord that thread is too deep for me to get into. I hate to ask for the Cliff Note version but its just too hard to get my head to wrap around it.

With that I'm still a little confused here.

Example situation:
Say I'm sitting there plucking at my bass that is plugged directly into my Vintech X73i and from there into my Apogee Ensemble.

What I am hearing is that when I look at the input chanel on Cubase 4, (for best/standard/good-rule-of-thumb results) I do not want the meter to be bouncing much above -10db?
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Old 2nd May 2007   #10
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Quote:
What I am hearing is that when I look at the input chanel on Cubase 4, (for best/standard/good-rule-of-thumb results) I do not want the meter to be bouncing much above -10db?
this is a good theorie.

other theorie:
record it relativly hot (I would not go over -4dB peaklevel) and mix it at lower levels ITB.


from the thread "PAUL FRIENDL"
Quote:
Every time this comes up I am left with exactly the same totally frustrated feeling.
The reason people do not get best results from ITB mixes and digital processing in general is that the whole cultural environment of metering, level control, overload and production styles within the digital domain is based on SAMPLE VALUE and not SIGNAL
However many times I re-itterate this very important fact it seems impossible for people to grasp exactly what it means and what the gravity of ignoring it actually is in repect of their audio results. And this is NOT even the user's fault, they cannot be expected to grasp it because they are totally buried in systems that are wholly based on sample value misconceptions and always display values which are NOT signal
For a really fair analysis, this is not primarily a user problem - it is an equipment problem that the user must make himself aware of if he is to avoid it.
People who hear differences are not wrong - the equipment is lying to you - it is ecouraging you to produce illegal results that you are not made aware of.
IMHO & LE this is the sole reason underpinning ALL the arguments about ITB mixing, sample rates, 'resolution' - you name it.
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Old 3rd May 2007   #11
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"New shit has come to light man" - The Big Lebowski
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Old 4th May 2007   #12
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Quote:
Originally Posted by RadioMoo View Post
  • The full resolution is there if you track at -18 at 24 bits; the old 16 bit thinking that you weren't using all the bits unless you tracked really hot does not apply here.

Exactly thumbsup

The dynamic range of a 16 bit file is about 96dB, the dynamic range of a 24-bit file is around 144dB! Even assuming cheap converters that "only" give you about a 20-bit practical effectiveness, you still have a good 20 or 30 dB of headroom before you have to worry about "losing bits".
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Old 5th May 2007   #13
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Hey guys... this knowledge has done wonders for my recordings. They no longer sound "harsh".

This is probably one of the single most important things I've learned here in GearSlutz land.
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Old 11th May 2007   #14
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weird weird weird

i've been reading up on these threads, and so for a recent project i tracked lower.
yes things didn't sound as harsh, and yes the mix was more open.
I turned up the monitor volume in the DAW and it sounded great.

But then I bounced the mix down to 16bit 44.1, and it sounded much quieter than the radio-friendly stuff. a few dB quieter.

perhaps this tracking lower doesn't work great for every style of music?
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Old 11th May 2007   #15
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Quote:
Originally Posted by cl516 View Post
weird weird weird

i've been reading up on these threads, and so for a recent project i tracked lower.
yes things didn't sound as harsh, and yes the mix was more open.
I turned up the monitor volume in the DAW and it sounded great.

But then I bounced the mix down to 16bit 44.1, and it sounded much quieter than the radio-friendly stuff. a few dB quieter.

perhaps this tracking lower doesn't work great for every style of music?
All the benefits you heard are exactly why you need to track a bit lower. the answer as to whether or not you got your mix as loud as commercial mixes has to do with the mastering process (I'm assuming your mix did not go to a mastering engineer). You can always stick a great sounding limiter on the 2-Bus to make the mix louder. It's the overall dynamic of each individual channel and how that plays into the gain staging of the mix that makes or breaks the end result.
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Old 24th May 2007   #16
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Just clued in to this post, what a brilliant little source of knowledge! Anyone have good recommendations for books on audio theory? I started off in Neuro Psychology and I can tell you how your ear perceives sound impulses but not a thing about the actual physics of acoustics. It's f***in fascinating though!

PiN
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Old 24th May 2007   #17
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I suspect a lot of the technical reason tracking lower sounds "ballsier" may be that the analog stages of converters and their power supplies aren't being pushed as hard. My personal method these days is to try and make sure that I error on the low side rather than on the high side of recording levels.
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Old 24th May 2007   #18
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Quote:
Originally Posted by rystro View Post
"New shit has come to light man" - The Big Lebowski
LOL - my favorite movie... exactly, this is great stuff, thanks for the advice.
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Old 24th May 2007   #19
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Try both way....record it kind a hot, and record at around -10 or -16 DB, and listen
to hear what sound best..........that's the way to do it.

All the best.
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Old 24th May 2007   #20
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read this:
PSW Recording Forums: Whatever Works => Digital tracking with low levels = better...is this new???
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Old 24th May 2007   #21
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There's no doubt that conservative tracking levels (into the converter) sounds cleaner. And usually better, but that's slightly subjective.

Some converters will gradually distort when getting close to the ceiling, I know one engineer who really likes that sound. Now, clipping a converter on purpose during mastering is something else I'm not adressing here.

As a rule of thumb I record at -6 dBFS to -3 dBFS in 24 bits and avoid getting any higher internally (plug-ins etc) in order to have some headroom. Not only are you avoiding some potential clipping issues but you can "corner" yourself with levels too high.

One example would be exceeding 0 dBFS output from a plug-in with fader at unity, and then lowering the fader. On the surface things would look alright (no clip) but the next plug-in e.g. a limiter would have it's input exceeded and you would have no real control over the amount of GR. It's easy to try for yourself.

But remember there's no sonic (or mathematical) difference in a floating point sequencer when lowering the faders and pushing up the master compared to raising the faders and lowering the master, as long as no clipping occurs. You can do a null test on this.
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Old 24th May 2007   #22
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One day, back in 2002, I was tracking vocals to a heavy metal band...
Sometime into the process I noticed the vocals sounding great, open, detailed, 3D...
When I looked into my waveform, it was smaller than I used to record, a good 6/8 db lower in amplitude.
After that, everytime I pay enought attention, I try to keep all my levels lower...
Now, I have a "real" explanation to that!
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Old 26th May 2007   #23
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Quote:
Originally Posted by rystro View Post
I've always been told, when tracking, that you want to get the "hottest" signal. So just make sure all signal/peaks are the biggest without clipping.

This is concerning rock/pop music.

Is this the correct way of thinking? Or do you want to give yourself some "headroom" on the channel?
maybe analog recording. This was due to tape hiss and signal to noise. But even with Analog
there are times when you don't want a hot signal. Big misconceptions with hot signal and analog.
Analog meters in the old days were slow. Even if a VU showed -3 it still could have been clipping cuz
the meter was slow to react. This is why plasma meters became popular later

With Digital tracking who cares you can always normalize it later.
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Old 28th May 2007   #24
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Quote:
Originally Posted by allencollins View Post
With Digital tracking who cares you can always normalize it later.
No, that's not good advice. Normalizing won't improve anything, to the contrary.

I've never seen plasma meters, maybe you're thinking of LED peak meters. But peak meters don't mean that VU meters are suddenly obsolete, they show two different things.
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Old 28th May 2007   #25
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Quote:
seen plasma meters
neve V3? had plasma meters..

I saw them going off and overheating a console..

funny
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Old 28th May 2007   #26
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Some SSL have plasma meters.... I think...
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Old 28th May 2007   #27
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my totally unscientific findings.

ive found tracking at -12 certainly the result seems to be better.
possibly because ADA conversion technology is still evolving ??
and will be for some time.
so for me...i'm tracking at tween -12 to possibly -8 now.
still testing the whole shebang out.
i can only postulate that there are various components within
a A to D circuit for example.
thus who knows how easy it is to overload one internal part ??
i tend to think of it like this. conceptually.
a river continues to move fast providing there are no impedences in its path.
mebe lower levels are best cos no possibilities of "problems/obstacles"
occurring in D to A circuits..particularly the cheaper ones ??
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Old 28th May 2007   #28
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I track with kick and snare peaking at -3 dbfs, bass, guitars and vocals at -6 dbfs. Mixing with the stereo buss peaking at -3 dbfs.
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Old 28th May 2007   #29
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The idea always has been about getting the signal to line level. That's what the mic preamps are for. That is the level that all the analog equipment in your signal chain works best at.

Line level is an average signal level (what an analog VU meter reads) not a peak level (what the meters in your DAW reads) or an average peak level (the average level of the peaks).

There is no standard for what a line level signal will equal in a DAW. Most of the converters that I have run across seem to be calibrated somewhere between -15dbfs and -20dbfs. The manual will normally tell you that spec, or at least tell you how much headroom it has so you can do the math. (18db of headroom would put line level at -18dbfs)

Sounds with low peak to average ratios like heavily distorted guitars, sine wave synth patches, etc... will have peaks around -15dbfs when recorded at line level (assuming line level = -18dbfs)

Sounds with high peak to average ratios like drums, piano, etc.. can have peaks up around digital clipping when recorded at line level.

Peak levels are only useful to see if you are clipping. The average level is the key to clean recordings.
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Old 29th May 2007   #30
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Quote:
Originally Posted by Lagerfeldt View Post
No, that's not good advice. Normalizing won't improve anything, to the contrary.
huh?

Anyone who worries about tracking levels in digital is clueless. As long as it's not clipping I DOESN'T MATTER.

Normalizing won't improve anything?? Do you know what Normalizing is??? Just curious.

VU meters are not acurate with spikey percussive instruments.
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