master fader in the red??
- Thread starter broken81
- Start date
LSD-Studio
HCAF crusher
xmarcelx said:
???
that does not have anything to do with the fact, that you're lowering the bit resolution when you lower the master fader.
24bit was just an example assuming you're bouncing 24 bit.
doesn't matter, that effects etc are more than 32 bit, when you bouce your stereo-output with24bit and it's at "0" ypo'll get 24 bit, when you lower it .....not.
mutant is right, though.
Dushan S
Member
You are not lowering bit resolution, if you are mixing inside the daw and everything is done inside daw. It seems that you are mixing bit resolution itself and what this bit resolution contains as data. And mixing bit resolution and volume.LSD-Studio said:
24 bit means that 24 bits are used to describe digital data, the number of bits is constant, and if audio is lower in volume for instance, less bits is used, yes, less bits to describe it, but it is not the same...
Sum of all audio chanels is in 32 or even 64 bits until it leaves daw AFTER the master fader. So what really counts is that everything is loud enough and comes close to zero at master fader so you are using maximum resolution when signal becomes 24 bit. In a similar way as it is good to have loudest possible signal when recording digital, without clipping. (if internal architecture of daw would be only 24 bit as it was on some older daws then it would be important not to go in red on audio channels and to mix with masterfader at 0 db, but it is not the case now, you can go over 0db on channels if it is 32 bit internal)
You are not able to have less than 24 bit signal in any way, it is just a matter what 24 bits contain. 24 bits is always there. It is a container, you can fill half glass of water but it is how much water is there, glass is still the same size. And you can never get all 24 bits used all the time, except all your audio tracks are recorded so that wav always peaks at -0.1 db and every tracks is compressed so it has minimum dynamics. And it would not be very musical. So those additional bits over "standard" 16 bits are mostly used to give higher dynamics while recording, and to give as better details of individual instruments.
It would be loss if you put master fader down and your highest peak is at -6.5db. If you normalise later +6db, it is NOT the same as if you were having -0.5db already when exporting from daw. In that case you are definitely losing resolution. It is similar like if you would work in photoshop on picture in 2400x1200 resolution (32bit internal data) and then export picture in 800x 600 and then put it back in photoshop and save 800x600 picture like 1200x 800 picture... You will get less quality picture than if you were exporting in 1200x800 in the first place. So to keep quality of audio while processing mixing etc, daws and plugins work internally with higher bit values than consumer audio or finished 24 bit mix. What you were saying could be true ONLY if master fader would be somehow AFTER the internal architecture of daw, like output fader of your soundcard (and then it would be important to feed maximum amount of data, maximum resolution to master fader, and it would be important to get faders up there but without clipping), but it is not the case, when you are exporting, resulting file is made after master section, masterfader. Hope I have explained it right, english is not my first language...
Kenneth R.
Cináed
misunderstanding... I thought you mean the fader at 0 not the level!LSD-Studio said:
And yeah Mutant is right!
LSD-Studio
HCAF crusher
Dushan S said:
you're right, basically that's what i meant, unfortunately english is not my native language
of course by bouncing 24bit you'll always have the 24 bit.
Noumenon
Obsidian Productions
So, the conclusion is that it is ok to slam the master a bit?
What about using a limiter at the master so you always have -0.3 ?
What about using a limiter at the master so you always have -0.3 ?
Dushan S
Member
I guess you better leave compressing shit out of the mix to mastering engineerNoumenon said:So, the conclusion is that it is ok to slam the master a bit?
What about using a limiter at the master so you always have -0.3 ?
You can use gentle limiting on groups or master fader to cut some peaks and get greater overall volume, but if you don't know what you are doing don't go too wild with that. There are some people that mix in master at the same time but not many of them. If you overcompress i a wrong way there is no turning back.
you're right, basically that's what i meant, unfortunately english is not my native language
We all doing our best are, do we 
sooooooooooo pull them faders down on tracks and leave master alone or pull the master down or just leave all alone and use a limiter to keep from peaking??
just seems like to me when i try and get my mix louder like with waves l2 it just distorts more and sounds like shit ???? is that because i ate up all my headroom by having master getting nailed already before i used limiter or what I'm just looking to get the most and have my final mix loud.
just seems like to me when i try and get my mix louder like with waves l2 it just distorts more and sounds like shit ???? is that because i ate up all my headroom by having master getting nailed already before i used limiter or what I'm just looking to get the most and have my final mix loud.
Mato Barla
Member
Mutant
I hate that supercow !
broken81 said:
AFAIK L2 is famous for being non transparent - if you want it to not distort your audio too much, increase ludness by ~4dB maximum.
My way of getting a nice loud mix/master (i'm doing both at the same time):
Use 32bit waves.
In Adobe Audition:
Normalize to 100%
Code:
Left Right
Min Sample Value: -18408.4 -29502.71
Max Sample Value: 20048.11 32765.98
Peak Amplitude: -4.27 dB 0 dB
Possibly Clipped: 0 0
DC Offset: -.006 -.012
Minimum RMS Power: -67.09 dB -61.62 dB
Maximum RMS Power: -4.75 dB -.98 dB
Average RMS Power: -19.63 dB -15.62 dB
Total RMS Power: -17.47 dB -13.45 dB
Actual Bit Depth: 32 Bits 32 Bits
Using RMS Window of -1003 msThere are usually 3 to 5 and all the rest is 3 to 5 dB lower.
Shave those peaks to the average level usually -3dB with a hard limiter.
Normalize to 100%
Code:
Left Right
Min Sample Value: -25991.87 -32767.64
Max Sample Value: 28307.08 32767.63
Peak Amplitude: -1.27 dB 0 dB
Possibly Clipped: 0 4
DC Offset: -.007 -.033
Minimum RMS Power: -64.09 dB -58.63 dB
Maximum RMS Power: -1.75 dB -.47 dB
Average RMS Power: -16.66 dB -12.66 dB
Total RMS Power: -14.51 dB -10.53 dB
Actual Bit Depth: 32 Bits 32 Bits
Using RMS Window of -1003 msUse L2 or any other limiter and be subtle if you want it to be as transparent as possible.
Statistics after L2 with -4dB threshold and -0,3dB ceiling:
Code:
Left Right
Min Sample Value: -31656.49 -31659.04
Max Sample Value: 31656.24 31658.66
Peak Amplitude: -.3 dB -.3 dB
Possibly Clipped: 0 0
DC Offset: .006 -.066
Minimum RMS Power: -61.22 dB -55.8 dB
Maximum RMS Power: -.77 dB -.77 dB
[COLOR="Red"]Average RMS Power: -13.51 dB -9.68 dB[/COLOR]
Total RMS Power: -11.47 dB -7.88 dB
Actual Bit Depth: 24 Bits 24 Bits
Using RMS Window of -1003 msIf you don't have Audition or any other host with "normalize to 100%" function you can use a freeware vst plugin Inspector - it will tell you exactly how much you are below or above 0dB - just enter the correct value in master fader and after rendering you will have a wave peaking at exactly 0dB without any clipping.
