Hi all,
I am in the process of sending tracks off to mastering.
Some of these mixes peak at around -3db and the master engineer wants them at -6db.
Is it ok to just fix it by turning down the master fader, or is that a problem in any way?
Using master fader for more headroom?
Hi,
You just need to leave the master fader at 0 and lower the gain knob for each track by a few db
You just need to leave the master fader at 0 and lower the gain knob for each track by a few db
I would strongly suggest otherwise - any/all "downstream" compressors used will behave differently when you change the input gain. Doing this will thus change your mix is possibly dramatic (and undesirable) ways!siln wrote:Hi,
You just need to leave the master fader at 0 and lower the gain knob for each track by a few db
There is absolutely nothing wrong with lowering the master fader for this purpose. Floating point audio systems don't care where you lower the gain, only that the final output is "legal" for the fixed point output (or you will clip). The only devices that "care" about level are non-linear devices such as dynamics and saturation/distortion, which of course will behave very differently with changes to input levels. Hope this makes sense…
Selig Audio, LLC
then the only problem is if you need to bounce it later for remix or mastering stems you will need to edit each track again , the thing with faders is they are the best spot for volume automation purpose and they are more sensitive at 0, that s the point i understand from the only touch fader at last. so maybe your faders are already automated in this case you would need to touch the gain knob or the selig re after compression/saturation. But as Selig said if it s ok soundwise turn down the master , i am not sure any inter samples peaks may occurs ? btw on that subject of gain staging , an intersample meter like the ssl x-isme as a re could be interesting
???Selig Audio, LLC
Why only touch the faders last? Plus, there is plenty of resolution across a wide range of the Reason faders - how much resolution do you need?siln wrote:then the only problem is if you need to bounce it later for remix or mastering stems you will need to edit each track again , the thing with faders is they are the best spot for volume automation purpose and they are more sensitive at 0, that s the point i understand from the only touch fader at last. so maybe your faders are already automated in this case you would need to touch the gain knob or the selig re after compression/saturation. But as Selig said if it s ok soundwise turn down the master , i am not sure any inter samples peaks may occurs ? btw on that subject of gain staging , an intersample meter like the ssl x-isme as a re could be interesting???Selig Audio, LLC
Again, it is a bad idea to adjust the gain knob late in the mix, for the reasons I've already mentioned.
Intersample peaks will be an issue no matter where you adjust gain, not sure what that has to do with the topic of achieving a little more headroom on the mix. The reason I suggest the master fader is because adjusting it will NOT affect any non-linear device, including the Master Compressor (if used). Depending on the routing of the Master Insert section, if it is set so that the Master Inserts are not "PRE COMPRESSOR" (as set in the master section on the Main Mixer), and you are using the Master Compressor for mixing, you could also get additional headroom by adjusting the Makeup Gain on the Master Compressor.
The main thing to observe is that if you change gain ANYWHERE "upstream" (before") a dynamics device, it can affect the way the device reacts and therefore change levels - and thus change your mix. The goal with adding headroom to your master is to NOT change the mix and simply add headroom!
You need to know your basic signal flow to understand what you can and cannot adjust without affecting devices down stream. But one thing is for sure: moving one fader down a few dB is always going to be MUCH easier than adjusting multiple gain controls, be they knobs or faders.
Selig Audio, LLC
Yes, and no… if there are no busses and no dynamics devices involved on the master, then yes it would be exactly the same thing. Being as it would be exactly the same thing, then the question becomes "why"?C//AZM wrote:Most def don't adjust the gain levels now. The master Fader will do the rick nicely but....
That little request to be able to move all faders simultaneously would perhaps fix the problem without moving the master fader.
Selig Audio, LLC
It's the old three ways to skin a cat,the mix sounds great but they're too high in the fader run,bring it all down by option selecting,many people have requested this,his mix looks silly like that,common practice so why ruin it even if it's not as crucial in this one case,easy way to do it,you just pointed out how it's not the same, trick.selig wrote:Yes, and no… if there are no busses and no dynamics devices involved on the master, then yes it would be exactly the same thing. Being as it would be exactly the same thing, then the question becomes "why"?C//AZM wrote:Most def don't adjust the gain levels now. The master Fader will do the rick nicely but....
That little request to be able to move all faders simultaneously would perhaps fix the problem without moving the master fader.
hey i am not a pro just so here are my 2 cents from what i have read and what i try to understand
this is something i tend to do , more and more and it works well for me , but i am also a lazy ass so lets say i dont follow this rule myself 100%
We dont know if OP has compressors channels, master comp or maybe a master limiter set at _0,5db
Follow these tips and you should give us the best start possible to optimising your music.
Try and keep peaks on the main mix bus between -10 to -4dBfs (Digital Full Scale) but no higher than -3 dBfs. If they are higher we would recommend lowering the individual mix element faders and group faders to reduce the level on the master output bus. You need to leave the master fader at 0 and work the faders and groups within the session (if Mixing in the box)
If mixing on a console please keep the digital capture recording levels conservative (-4dbfs peak) suggested.
The reason to do the above (both in the box and recording the console output) is not to do with headroom for mastering, (mastering is carried out in the analog domain so headroom is irrelevant). It’s because of intersample peak distortion which starts to happen as low as 4 db below dfs in some computer workstations. When digital mixes have intersample distortion it becomes very hard to get a good clean result from analog hardware as it responds badly to these peaks that your software meters cannot even detect as they happen between sample cycles. Working natively at higher sample rates does a lot to alleviate this issue but only if the parts in the session were recorded at that sample rate. Do not upsample files under any circumstances. You are better off at 44.1K if that is what your session was originally.
I dont know much about intersample distorsion , got ozone and i know it suppress it , but if i understand it is sometimes better to avoid them before having to suppress them? , just discovered the wave plugin , seems over consumming dsp, maybe it could be handful as a Re...just saying
Since -3 db is not sufficient for his engineer , maybe he is more affraid about distorsion than headroom (if hardware master)
I am not arguing with you , and hopefully you are here because my fist post was a little fast but i assume you will agree that this is better to fix things at first than having to do weird thing to correct them, maybe it is the nerds debate about keeping things organized , the OP can lower his master , i just wanted to warn him than it can be handy to correct his mix as the engineer wanted for not being stuck again in the future , yes it is easy now, today , but the next week maybe he will have to get back to the mix channels individually anyway because his engineer offers him stems mastering or for remix purpose.
Because they are the most handy spot for automation levels and that they have not the same resolution at 0 than at -12 or 12 ,try to handle control over the gain since start , set the gain knob to his level , set your synth channel to the right level, before it hits your compressor, then control the gain out of the comp channel and so on...in this process the channel fader is the last to "set" , but if you already set all lvls before you dont need to touch it anymoreWhy only touch the faders last?
this is something i tend to do , more and more and it works well for me , but i am also a lazy ass so lets say i dont follow this rule myself 100%
yep that is why it is better to start gain staging before the channel fader and before the inserts chain , if a comp then just turn the out off, otherwise they wouldnt make an audio out knob for compressorsAgain, it is a bad idea to adjust the gain knob late in the mix, for the reasons I've already mentioned.
We dont know if OP has compressors channels, master comp or maybe a master limiter set at _0,5db
Got this from here : http://www.stardeltamastering.com/pre-mastering-advice/ : General Mix Levels and ProcessingIntersample peaks will be an issue no matter where you adjust gain, not sure what that has to do with the topic of achieving a little more headroom on the mix
Follow these tips and you should give us the best start possible to optimising your music.
Try and keep peaks on the main mix bus between -10 to -4dBfs (Digital Full Scale) but no higher than -3 dBfs. If they are higher we would recommend lowering the individual mix element faders and group faders to reduce the level on the master output bus. You need to leave the master fader at 0 and work the faders and groups within the session (if Mixing in the box)
If mixing on a console please keep the digital capture recording levels conservative (-4dbfs peak) suggested.
The reason to do the above (both in the box and recording the console output) is not to do with headroom for mastering, (mastering is carried out in the analog domain so headroom is irrelevant). It’s because of intersample peak distortion which starts to happen as low as 4 db below dfs in some computer workstations. When digital mixes have intersample distortion it becomes very hard to get a good clean result from analog hardware as it responds badly to these peaks that your software meters cannot even detect as they happen between sample cycles. Working natively at higher sample rates does a lot to alleviate this issue but only if the parts in the session were recorded at that sample rate. Do not upsample files under any circumstances. You are better off at 44.1K if that is what your session was originally.
I dont know much about intersample distorsion , got ozone and i know it suppress it , but if i understand it is sometimes better to avoid them before having to suppress them? , just discovered the wave plugin , seems over consumming dsp, maybe it could be handful as a Re...just saying
Since -3 db is not sufficient for his engineer , maybe he is more affraid about distorsion than headroom (if hardware master)
I am not arguing with you , and hopefully you are here because my fist post was a little fast but i assume you will agree that this is better to fix things at first than having to do weird thing to correct them, maybe it is the nerds debate about keeping things organized , the OP can lower his master , i just wanted to warn him than it can be handy to correct his mix as the engineer wanted for not being stuck again in the future , yes it is easy now, today , but the next week maybe he will have to get back to the mix channels individually anyway because his engineer offers him stems mastering or for remix purpose.
I have long recommended, and the Props seem to suggest the same with the user guide and FSB patches, that you keep all audio channels around peaks of -12 dBFS, and your mix between -6 to -3 dBFS. One problem in Reason is that the channel meters in the Main Mixer do not show Peak level, but instead show RMS. There are ways around this, one is to solo the track and use the Big Meter (making sure to bypass ALL master processing first). It is also the reason I created the Peak Hold meter in Selig Gain. I have also recently pleaded my case with the Props, to simply have the channel meters follow the meter mode of the Big Meter (hopefully they will consider this in a future update).
As for the fader "resolution", I will again ask how much resolution do you need? The answer will come in the form of another question, which is "how small of a change in level can you hear?".
For example, you say there is less resolution at -12 on the fader, but I say that there is 0.06 dB resolution on the fader at -12 dB. How often do you make changes of 0.06 dB in your mix? Personally speaking, I can hardly perceive changes of 0.25 (over four times the fader resolution at -12 dB), and often don't make changes of less than 0.5 dB if that!
So if 0.25 dB is more than enough resolution for most mixing, how far down do you think you can lower the fader and STILL have that much resolution? The answer may surprise you, because it is when the fader is almost touching the bottom, below -50 dB, that the resolution falls below 0.25 dB!!! For WELL over 90% of the fader travel you have over 0.25 dB resolution. For around 50% of the fader travel you have 0.01 dB fader resolution!
This is one of those things like sample rate, where folks assume you ALWAYS want MORE. But just like sample rate, there is a point where the increases don't actually give you anything "more". And that is the case with fader "resolution", where you really need to access your NEEDS and then you will realize that 90% of the time there's no loss in moving the channel faders where ever you need them - and where the mix sounds "right".
As for the fader "resolution", I will again ask how much resolution do you need? The answer will come in the form of another question, which is "how small of a change in level can you hear?".
For example, you say there is less resolution at -12 on the fader, but I say that there is 0.06 dB resolution on the fader at -12 dB. How often do you make changes of 0.06 dB in your mix? Personally speaking, I can hardly perceive changes of 0.25 (over four times the fader resolution at -12 dB), and often don't make changes of less than 0.5 dB if that!
So if 0.25 dB is more than enough resolution for most mixing, how far down do you think you can lower the fader and STILL have that much resolution? The answer may surprise you, because it is when the fader is almost touching the bottom, below -50 dB, that the resolution falls below 0.25 dB!!! For WELL over 90% of the fader travel you have over 0.25 dB resolution. For around 50% of the fader travel you have 0.01 dB fader resolution!
This is one of those things like sample rate, where folks assume you ALWAYS want MORE. But just like sample rate, there is a point where the increases don't actually give you anything "more". And that is the case with fader "resolution", where you really need to access your NEEDS and then you will realize that 90% of the time there's no loss in moving the channel faders where ever you need them - and where the mix sounds "right".
Selig Audio, LLC
OK! then the resolution is fine you are right it is fun because it was the only argument i had that didn t come from my personnal experience, i never had to complain about it personnaly , nevertheless if u suceed in proper gain staging before your faders, you dont have to touch them anymore, and they are free now for the automation levels; the thing is I assume OP got it right at -3 db , but the engineer is touchy about this and prefer -6 db, maybe for distortions purpose more than headrooms, it s not that big error as clipping master but i assume the engineer consider -3db as clipping ; from my personnal point of view i would trust an engineer that prefer -6 than -3 and giving him a proper mix , with the master set at zeros if i never touch the master i wont touch it for this only purpose , i will do the Mix thing ; )
YEs , that s a big thing and i think we all need at least peak meter on the channels for this purpose and maybe a kind of xintersample meter!I have long recommended, and the Props seem to suggest the same with the user guide and FSB patches, that you keep all audio channels around peaks of -12 dBFS, and your mix between -6 to -3 dBFS. One problem in Reason is that the channel meters in the Main Mixer do not show Peak level, but instead show RMS. There are ways around this, one is to solo the track and use the Big Meter (making sure to bypass ALL master processing first). It is also the reason I created the Peak Hold meter in Selig Gain. I have also recently pleaded my case with the Props, to simply have the channel meters follow the meter mode of the Big Meter (hopefully they will consider this in a future update).
Jumping in with a question just because I'm curious: (It may be the stupidest question of the month, but still): Would it be a catastrophe to just send it as it is?selig wrote:I would strongly suggest otherwise - any/all "downstream" compressors used will behave differently when you change the input gain. Doing this will thus change your mix is possibly dramatic (and undesirable) ways!siln wrote:Hi,
You just need to leave the master fader at 0 and lower the gain knob for each track by a few db
There is absolutely nothing wrong with lowering the master fader for this purpose. Floating point audio systems don't care where you lower the gain, only that the final output is "legal" for the fixed point output (or you will clip). The only devices that "care" about level are non-linear devices such as dynamics and saturation/distortion, which of course will behave very differently with changes to input levels. Hope this makes sense…
If the mastering guy needs more headroom, they could turn down their slider after importing it?
I guess that's a tiny fraction more noise since there's an additional rendering, but depending on the resolution of the thing that the engineer receives it feels like neglible?
i would assume at this point if an engineer wants -6 db instead of - 3 db , it s that he cant /dont want do to himself as the rendering at -3 is too much for him ; from what i understand if you send the rendering - 3 db it wuld stay -3db at the degree of saturation and everything else , that is why he wanted the artist to render at lower db
but shouldn't the mastering engineer be applying different amounts of saturation, compression and limiting for each track? it sounds a little suspicious to me. he should be able to work with any track as long as it isn't clipping. -6dB is just a reference for mixing but it holds no importance for the sound. is he just throwing the track through a preset and charging for it?siln wrote:i would assume at this point if an engineer wants -6 db instead of - 3 db , it s that he cant /dont want do to himself as the rendering at -3 is too much for him ; from what i understand if you send the rendering - 3 db it wuld stay -3db at the degree of saturation and everything else , that is why he wanted the artist to render at lower db
some very good engineers like the ones from http://www.stardeltamastering.com/links/ want their masters to be lower than -4 db, it should let them more control over the hardware , it s like the less db you give them , the more control they got and the more they are happy , because if there is less, they can always turn up the volume, but if there is more , they cant do anything, you alreadyy crossed the line of distorsion and they cant go back since you "rendered" them a mix that has been converted in D/A/AD digital to analogique
Hmm...this is where I don't understand. What would be the problem with them turning down the volume compared with if it's already turned down when they receive it?siln wrote:as said earlier, some very good engineers like the ones from http://www.stardeltamastering.com/links/ want their masters to be lower than -4 db, it should let them more control over the hardware , it slike the less db you give them , the more control they got and the more they are happy , because if there is less, they can always turn up the volume, but if there is more , they cant do anything
the problem will stand in eventual clipping/distort/saturation of sound , that means the lower level they want to avoid these problem that are distortions and clippings occurs as soon as -4 db
that means you can lower your mix as much as you want , and mix at -20 db you will have no problem , just turn on the volume at the end , but as soon as you reached the point of too high , there is no going back and engineer cant do anything , well he could correct things but a good engineer isnt supposed to correct things he just want to improve it
Heh I was confused on different levels - I realize that we are not talking about peak level here right? So the extra headroom they want is to reduce the risk that there is distortion already in what the mastering engineer receives?siln wrote:that means you can lower your mix as much as you want , and mix at -20 db you will have no problem , just turn on the volume at the end , but as soon as you reached the point of too high , there is no going back and engineer cant do anything , well he could correct things but a good engineer isnt supposed to correct things he just want to improve it
i think yes; it s like saying -3db = 0 db , it s the same shit, because shits happens at this levels, so lower the shit out
From what i learned so far , the much your lower your levels , the more engineers is ok ith yu !
siln wrote:that means you can lower your mix as much as you want , and mix at -20 db you will have no problem , just turn on the volume at the end , but as soon as you reached the point of too high , there is no going back and engineer cant do anything , well he could correct things but a good engineer isnt supposed to correct things he just want to improve it
You apparently have never heard of quantization errors.siln wrote:From what i learned so far , the much your lower your levels , the more engineers is ok ith yu !
Levels in digital audio can't be too low either, or else the number of bits will start lacking to properly encode the audio and the consequences can be just as bad as clipping. Increasing the bit depth from the start greatly helps reducing that risk of course, but this particular aspect of digital audio still shouldn't be neglected.
Doesn't happen with floating point. You always have 24-bits of precision in the mantissa, the exponent changes as the scale of the values change. Floating point is very well suited for PCM encoding. The size of the steps between values get smaller as the values themselves get smaller.odarmonix wrote:You apparently have never heard of quantization errors.
A crude example would be values from 0.1 to 1 get 10 steps:
0.1, 0.2, 0.3, 0.4, 0.5, 0.6, 0.7, 0.8, 0.9, 1.0
When you go up a level magnitude from 1 to 10, you still get 10 steps:
1, 2, 3, 4, 5, 6, 7, 8, 9, 10
And one more, from 10 to 100:
10, 20, 30, 40, 50, 60, 70, 80, 90, 100.
Except you actually get 16 million steps at each scale level, and there are 256 scales.
-
- Information
-
Who is online
Users browsing this forum: Popey and 25 guests
Share and download patches!
Join ReasonTalk on Slack!