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Huh? Sliders change the levels of the track that 'get sent to the mix'. That is how both digital and analog work. When you reduce the slider you reduce the signal out to the mix (or more exact, out from that track); what you hear in your monitor mix IS your mix. It isn't compression or limiting that does it. It's a reduction in the volume for the track output. It's digital.
Ok, fair enough. That seems to make intuitive sense with one problem. Why then do recording engineers universally preach do your basic tracking at -12 to -18db? And in the case of RB, tracking includes generating an RT. Who cares about that if all you have to do is lower the faders during mixdown? Obviously that is not the proper way to do it or all those engineers would simply say that and not go on and on about recording levels.
Like I said, I don't have the education to explain this fully but I do know it's true. Unlike some people I know when I read the same thing written over and over by pros and I finally conclude they know what they're talking about I simply try to do it the same way. I don't have two years to study it to try to figure out all the why's. I just do it.
You do pose a good question Rharv. It's doesn't make obvious sense and if someone here can answer it in detail I would like to read it. You could record every track at -2 and just adjust the console faders but every pro says no, don't even think of doing that.
Bob
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"Cleanup in aisle 9, stat!"
Lots of terminology is starting to be thrown around here. One of the items that is likely being misinterpreted is the term introduced by jazzmammal as 'automatic limiting' when saving the 2 track mix to a stereo .wav.
In most DAW software, when one does a saving of the final output bux to a 2 track file, there is an option for 'normalizing' or 'normalize' or 'prevent clipping' or something along those lines.
What jazzmammal is describing is a normalizing process. What normalizing will do is look to see where the absolute maximum level would be in the samples of the mixdown, and set that so that particular sample's digital output value is 100% of the max possible digital output - coded to .wav - and then scale every other sample in the mixdown by the same scale factor.
It can be a greater than 1 multiplication, or less than one (as is being discussed in this thread) multiplication.
This is entirely different than automatic 'limiting', which looks to see if any part of the mix-down file would clip and reduces only those parts which would clip, leaving the others unchanged.
Does everyone understand the difference? Normalizing scales every single sample in the .wav file by the same amount, in order to achieve the end result that the absolute loudest sample in the mixdown, gets the max that the .wav file allows, where limiting only changes samples which exceed some threshold value, leaving other samples unchanged.
Are we clear? If not, please post as such and I'll try to explain in a different way.
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...automatic limiting ...
I am confused by this term. It sounds like you are implying that the DAW software removes clipping automatically on mix down with no intervention by the user. Does this happen in RB? I don't think Sonar does that (I could be wrong, of course).
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Scott just answered that. As I was writing these posts something didn't sound right and this is it, it's not limiting, it's normalizing. Thanks Scott. Just switch those two words. My main point hasn't changed and now when I search mixing articles using the keywoard "normalizing" I'm getting lots of hits all saying versions of this:
Normalizing
Do not use any kind of normalizing on your mixdown.
Normalizing raises the signal level in an unnecessary fashion, and it will change the
amount of headroom left in the mix. The final volume level of the mix will be optimized by
me.
That's what I was missing. I'm not going to bother posting the full link, they all say the same thing. Pro's don't normalize in most cases so my overall point is correct.
Bob
Biab/RB latest build, Win 11 Pro, Ryzen 5 5600 G, 512 Gig SSD, 16 Gigs Ram, Steinberg UR22 MkII, Roland Sonic Cell, Kurzweil PC3, Hammond SK1, Korg PA3XPro, Garritan JABB, Hypercanvas, Sampletank 3, more.
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Pros are often feeding signal to other hardware equipment. In this scenario headroom is important. If you are staying in the digital realm, you are a little safer going up a bit on the levels. Besides headroom there is signal-to-noise. Many hobbyists focus on this aspect. Numerous tracks add noise and if you record a strong signal you can lower signal-to-noise, but at the cost of headroom. I'd worry more about whether I was using line levels that are consumer (-10) or pro (+4). That may also come into play for the reason 'pros' say -12; they are likely using signals with more energy than hobbyists are to begin with. Here, maybe this will help- http://www.independentrecording.net/irn/columns/jwal/index.php?id=79
I do not work here, but the benefits are still awesome
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You CANNOT mix down PROPERLY just by moving the volume sliders on your mixing console or in this case RB's software mixer. That is merely for you to adjust your monitor mix, it's not physically changing the levels on each track.
Wow, this has to be the most ill-informed quote I have ever seen on this forum....just....wow....
So I guess I would have to adjust each tracks waveform to do a PROPER mix?
I think someone needs to buy some books & study some more....wow..... 
Last edited by jcspro40; 10/28/12 04:05 PM.
i5 3.20GHz, 32gb RAM, 1tb SSD OS, 12tb HDD, 4gb gForce vid card, 32" monitor, Audient id44, Win10 x64, BiaB/RB 2023, Reaper 6,IK Multimedia Total Studio 3.5 MAX, Waves 10
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Quote:
Quote:
Quote:
You CANNOT mix down PROPERLY just by moving the volume sliders on your mixing console or in this case RB's software mixer. That is merely for you to adjust your monitor mix, it's not physically changing the levels on each track.
Wow, this has to be the most ill-informed quote I have ever seen on this forum....just....wow....
So I guess I would have to adjust each tracks waveform to do a PROPER mix?
I think someone needs to buy some books & study some more....wow.....
What the poster was referring to is that the sliders are not performing a destructive edit to the levels of the recorded track.
There have been more misinformed statements in the thread than this one.
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As analogue levels and digital levels are not the same, I thought this article on recording levels might be of interest. http://www.massivemastering.com/blog/index_files/Proper_Audio_Recording_Levels.php There seems (at least to me), to be two different things being discussed in this thread:the recording levels (how hot to record the tracks) and the slider levels on the mixer. I'm trying to learn how to do this stuff properly and was getting confused so I thought I'd post the link and see what everyone else thought. 
yjoh
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Pretty good article, but of course he doesn't say why it is harder to make hot tracks loud(er) while it is easier with track with lots of headroom.
My take: You use some kind of compression/limiter to make things loud at the end. If everything is too hot, then the "brick wall" limiter does all the work and everything gets squashed and you lose dynamics and clarity. If you have headroom, then the compressor does most of the work and the limiter only kicks in at a few transient peaks.
Well, I think that is what happens. But I don't really know for sure. I mean you could just slide the faders down and "create headroom" -- but I don't think that works as well. Ya know, I think I have hit the 0dB of my knowledge on this subject.
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Quote:
Pretty good article, but of course he doesn't say why it is harder to make hot tracks loud(er) while it is easier with track with lots of headroom.
My take: You use some kind of compression/limiter to make things loud at the end. If everything is too hot, then the "brick wall" limiter does all the work and everything gets squashed and you lose dynamics and clarity. If you have headroom, then the compressor does most of the work and the limiter only kicks in at a few transient peaks.
Well, I think that is what happens. But I don't really know for sure. I mean you could just slide the faders down and "create headroom" -- but I don't think that works as well. Ya know, I think I have hit the 0dB of my knowledge on this subject.
I DON'T user compression/limiter until the very last step if at all.
Think about this. If you have a quiet piano track, once you pull that slider all the way up or right (depending on your screen view) you are done. That's as loud as you can go. That's where I would start boosting 2db at a time to see if that created enough headroom.
And that is why I prefer to record hot. I can slide that puppy down to nothing if I want to, but I can only go so high before there is no more "high" to go to.
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... Think about this. If you have a quiet piano track, once you pull that slider all the way up or right (depending on your screen view) you are done. That's as loud as you can go. ...
In sonar 8.x you have four ways to crank the gain on a single track (and there might be a couple others!). Of course clipping is your enemy with these steps.
1.) push the fader all the way (maximum +6db, it looks like)
2.) Crank the trim control (max +18db)
3.) Just add gain to the track. In Sonar 8x, you highlight that track, go to process --> audio and add max of +18db as many times as you want.
4.) You can use compression with make-up gain to make it louder.
It looks like you are just using the fader.
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I am also not in Sonar 8.x. I work in Real Band 95% of the time. I go to Sonar when I want to play with an effect that RB doesn't like (Izotope, for example, crashes my RB but works fine with Sonar).
The original question, 3 pages ago, was asking what people call mastering, and why some of my songs sound like the top half of the spectrum is washed out. Early on I got like 4 different thoughts as to what the definition os mastering is. The resulting conversation has been awesome on many levels, too. Links and links from those links and all good reading that I copied and printed for highlighting.
This is why I love this place!!!
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Just checked -- In Realband you seem to have everything that Sonar does.
1.) The fader is the same.
2.) The trim is found by clicking the fx icon on each track (next to the Solo button). The trim control is in the lower right of the fx window.
3.) To add gain to a track: Right-click on a track and "Select Whole Track". Then Edit menu --> Audio Effects --> Gain Change
4.) And I think you know how to find the compressor.
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Eddie, try using DXi version of Ozone. Works here. If that's what you have running that crashed RB then try the VST version. I've heard of both ways having success, so worth trying 'the other one'.
Oh, and if you find yourself boosting lots of tracks, consider pulling the other tracks down instead.
Then worry about final levels (mastering) later. After the mix is done.
I do not work here, but the benefits are still awesome
Make your sound your own!
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Hi Kevin,
Referring to your quote:
"My take: You use some kind of compression/limiter to make things loud at the end. If everything is too hot, then the "brick wall" limiter does all the work and everything gets squashed and you lose dynamics and clarity."
Your knowledge isn't 0db at all, from eveything I have read and studied, you have it right.
If the wav is already filling the entire window, you have no more room to add effects and the mastering engineers are not going to like it. They have nothing to work they magic on. Pulling faders up or down doesn't alter the wav itself, it doesn't give you more headroom (as referred to in the article),it only the changes the volume.
At least that's my understanding of it all.
yjoh
[i]Music...a joy for life.
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Ya know, I think I have hit the 0dB of my knowledge on this subject.
Now THIS made my day!!!!  
i5 3.20GHz, 32gb RAM, 1tb SSD OS, 12tb HDD, 4gb gForce vid card, 32" monitor, Audient id44, Win10 x64, BiaB/RB 2023, Reaper 6,IK Multimedia Total Studio 3.5 MAX, Waves 10
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If the wav is already filling the entire window, you have no more room to add effects and the mastering engineers are not going to like it.
I hate to keep disagreeing with people, but this isn't strictly true. (Or even at all.)
Suppose the mastering engineer is sent a wav file with very little dynamic range and the recording normalised (maxed out). Because this is digital, the first thing he does is make a safety copy to work on. He then performs a gain change down a few db to create headroom without effecting the original mix in any other way. He is then free to work on the track in whatever way he sees fit, including increasing the dynamic range by use of an expander. When he's finished, he can bring the whole mix back up to max. Job done.
ROG.
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Quote:
Quote:
If the wav is already filling the entire window, you have no more room to add effects and the mastering engineers are not going to like it.
I hate to keep disagreeing with people, but this isn't strictly true. (Or even at all.)
Suppose the mastering engineer is sent a wav file with very little dynamic range and the recording normalised (maxed out). Because this is digital, the first thing he does is make a safety copy to work on. He then performs a gain change down a few db to create headroom without effecting the original mix in any other way. He is then free to work on the track in whatever way he sees fit, including increasing the dynamic range by use of an expander. When he's finished, he can bring the whole mix back up to max. Job done.
ROG.
But I bet he (or she) wouldn't be a very happy mastering engineer (unless he got to charge a lot for his services). Fixing a bad mix, while in the realm of a mastering engineer, is probably the last thing he would want to do to try and make it sound as good as possible.
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When I listen to your songs Eddie it sounds like they have a poor job done on the EQ. That may be the problem with sounding washed out on the high end.
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Yep. Exactly what I am trying to learn. Plus some of the earlier songs were recorded through an outboard mixer with outboard reverb and EQ and I need to do them all again without all that stuff. Once those destructive effects were recorded, there is no going back, so I need to sing some of them again. First I have to hope I still have the original SEQ files.... it'll come. With all the great input from the userbase here, it'll come. The later stuff is MUCH better as I have learned more.
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We've just released XPro Styles PAK 9 for Windows & Mac Band-in-a-Box version 2025 (and higher) with 100 brand new RealStyles, plus 29 RealTracks/RealDrums!
We've been hard at it to bring you the latest and greatest in this 9th installment of our popular XPro Styles PAK series! Included are 75 styles spanning the rock & pop, jazz, and country genres (25 styles each) that fans have come to expect, as well as 25 styles in this volume's wildcard genre: funk & R&B!
If you're itching to get a sneak peek at what's included in XPro Styles PAK 9, here is a small helping of what you can look forward to: Funky R&B Horns, Upbeat Celtic Rock, Jazz Fusion Salsa, Gentle Indie Folk, Cool '60s Soul, Funky '70s R&B, Smooth Jazz Hip Hop, Acoustic Rockabilly Swing, Funky Reggae Dub, Dreamy Retro Latin Jazz, Retro Soul-Rock Fusion, and much more!
Special Pricing! Until July 31, 2024, all the XPro Styles PAKs 1 - 9 are on sale for only $29 ea (Reg. $49 ea), or get them all in the XPro Styles PAK Bundle for only $149 (reg. $299)! Order now!
Learn more and listen to demos of XPro Styles PAKs.
Video: XPro Styles PAK 9 Overview & Styles Demos: Watch now!
XPro Styles PAKs require Band-in-a-Box® 2025 or higher and are compatible with ANY package, including the Pro, MegaPAK, UltraPAK, UltraPAK+, and Audiophile Edition.
Video: Band-in-a-Box® 2025 for Mac®: VST3 Plugin Support
Band-in-a-Box® 2025 for Mac® now includes support for VST3 plugins, alongside VST and AU. Use them with MIDI or audio tracks for even more creative possibilities in your music production.
Band-in-a-Box® 2025 for Macs®: VST3 Plugin Support
Video: Band-in-a-Box® 2025 for Mac®: Using VST3 Plugins
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