Decibels vs volume

Lots of people have opinions on this.
- EBU Meter (European Broadcasting Union) measures loudness over time. Just decibel is not enough... pitch is also taken into account.
- VU meters have 'ballistics' or slow response both up and down that makes them sort of a rolling average where more recent data is considered more than older data.
 
When I first started, I would mix below 100%, then bounce and normalize the track to see where the loudest moments are and go back and adjust levels in the mix to dip those levels then re-bounce.
After a couple of bounces, I'd apply some limiting to add a dB or two to the volume.
But this isn't a good way to get different tracks sounding with the same loudness on a CD.
 
When you use an EBU meter (what I do now), I reduce compression and limiting on louder tracks to push down the relative volume and increase it on softer tracks to bring up the volume. This gets you an average volume across the CD and also helps you set compression as minimally as you can to get to the volume you are targeting.

There is a long process to master a CD, I use Wave lab to master and it has special tools for this. 
First learn to use your ears, not meters. Over time you will learn to hear on your monitors the correct levels for things like bass. Visual aids will help you learn, but your ears do get better the more you practice mixing. 
 
But here's a simple solution using what you already have in Sonar.  There are many ways to do this but I'm offering what works for me for the last 10 years. This is a visual aid. 
 
Put the LP 64 multi band compressor on your master buss. It will show you instantly which frequencies are pushing your master output. Learn how to use it.  
  
And on the topic of Decibels. Your meters ( Sonars) only show you peaks. A song could have one snare hit and peak @ odb. You can manually look for these or you can use a compressor to bring everything else up. Each method has it's pro's and cons. Manual can give more natural results but it super time consuming. Compression can trash a songs vibe but this is depending on the type of music. I happen to like just the right amount on my music. 
 
So you export your song from Sonar, you used the Multi band ( or??) and it's sitting at -.5db but it's still not as loud as a commercial release.?? 
The next move is to use a global analyzer to show you what the songs average RMS level is. This tells us how loud the song really is.  The modern benchmark depending on style of music is around -14db. . Once again you need to know how to use these tools and use with your ears wide open. There are other tools taht show average RMS, some cost a lot of money. 
 
 
 

The first thing to do before burning a CD is to do an export of your song @ 16bit 44.1KHz
 
Import it back into the same project onto a new track. Route this track to a new bus and route this straight to your main outs and compare it to what your mix sounds like
 
Solo your new bus and quick group it with your existing master buss. now you can easily switch between the import and the mix just by pressing the solo button on/off
 
If there is no perceived difference between the 2 then great - you can use the 16 bit export to burn your CD with. 
 
So your project will look something like this:
 
Tracks > Busses > Master Buss > Main Outs
 
Imported Track > New Buss > Master buss > Main Outs.
 
 
If there's anything wrong with the routing of your project this will be immediately apparent. (hint : muting your Master Buss should result in total silence. If it's not, you've got something going straight to your Mains, thereby byapssing any Fx on your master buss)

Lots of good descriptions and explanations above.
 
Here's a cheat. Pros may facepalm but for rough-mastering so I can listen to stuff on the go with a minimum of effort this works well for very little time investment.
 
It doesn't matter whether you do this on the maser bus or export your finished track to a .wav file for import into your favourite product. Sound Forge Pro works well but really it could be anything. Personally I prefer to export because sometimes I want the original undiluted version for comparison or for someone else to work with.
 
1) Load your exported .wav into app, or cackle like and end-of-level boss and go to the master bus.
2) Apply LP64 Multiband. The exact settings will depend on project. Occasionally I use the default Softer or Louder mastering setting but because I tend to mix quite quietly I have a custom FXP I created with the settings just how I like them. You may always religiously create tracks that are close to -6Db all the time and create a custom FXP for that expected output. Whatever works for you.
3) Boost. Make sure the output is set to Zero DB, and not the default of -3. I refer here to the Cakewalk Boost plugin and not to a chocolate bar.
4) Save.
5) Export to MP3 if required.
 
Done.
 
This is not pro mastering, this is nowhere near as good as pro mastering, it is not meant to replace pro mastering. This is just a quick and dirty way to get something to listen to ambling around the countryside or whatever you do when not being awesome in front of your DAW.

May I call you Jeff?
 
I say to you, as a self described newbie, there is no plug in until you know what is does.
 
As I learn what the "workman like" basics of signal processing (for me "Dynamics and EQ"). I get better at knowing what I am looking for does. 
 
The thing that has done more for my understanding (I am 4 year old or so) was the introduction of the LM Comp and EQ.
 
I love the head phone icon. It is a great teacher. Create just one band in either tool, slide it around (sweeping) thru the signal while you listen, and listen to what you are limiting or adding (headphone button "no headphones required" it is "listen").
 
Oh by the way the first thing you should do is set those babies to non-linear. They can be nasty in a crowd in the higher settings.
 
This method gives visual data and audible clues. By the way I think it took me 3 years to even have an MP3 I would let someone play on their phone.
 
AMK

To add to my previous post and the others.  Don't use the visual feedback as the main criteria for your decisions.
 
Let your ears do the work, not your eyes.  If it has too much bass, turn it down, not enough, turn it up.  Or contour it so it fits better.  It can sound like too much because the space isn't carved out.

Does the resulting CD sound OK elsewhere?  If it sounds OK on your PC and if you put the tunes on your phone or in another car and it is OK, the issue is your car.  Some cars have "Bass Enhancement" or loudness enhancement that attempts to give people the home environment on the road.  I personally don't like all the extra bass this puts in.
Just my 2 cents...

patm300e
Does the resulting CD sound OK elsewhere?  If it sounds OK on your PC and if you put the tunes on your phone or in another car and it is OK, the issue is your car. 

 
That's a really hasty conclusion to come to. The job of mixing is to make mixes that sound good in all kinds of non-optimal playback situations.
 
 
 
 
 
 

Man, I really appreciate everyone's help on this! Again, this forum is truly the best around. 
 
So many good ideas here that I think will really help me, and others in my position. I've finally got a mix that's been gain staged correctly, unlike my previous attempts. But it always seems to be the freaking low end that becomes my nemesis. I'm not really trying to master anything yet, but just wanna get a decent rough mix that proves I'm on the right track. I'll definitely try the song on other devices, instead of my car. I think that may be an issue. One good thing I did notice though......the "mud" factor has been significantly lessened on this mix compared to previous ones. So at least my EQ'ing is getting better (thank you HPF). 
 
I am using my ears, not just the meters. Actually, I'm using both complimentary, I think. So I'll keep at it and report back. I am gonna buy some acoustical treatment for my room this week. So I think that will be a big help.
 
Thanks again guys.
 
~Jeff

You have to use both your ears and your meters. The problem is every studio (space)  sounds different. Especially in loe frequencies. 
In my studio the bass my sound small while the same volume setting on the same bass sounds huge in your studio. Finding the right  bass volume is definatly a realy hard task. Thats why i am so glad about the dbvu trick. It is not a rule but with this method you can be sure you are in the ballpark.
 
And this little trick works very well with the mentioned headphones. Headphones are even worse mixing bass than monitors. Littel exapmple. Ever heard the Bass on some AKG 701 and than some for example beyerdynamics or even beats? XD Mixing bass on headphone by ears is a task that cant be accomplished as long as you dont use your headphoens for years and you know them really really well.

Tosmurf
I have learned i really nice "trick" from Grammy Winner Jacquire King.
He wrote this in an article years ago. Its about getting your bass allways to the right volume. Dosent matter where you mix (other room other sound...).
 
I personally use this in every mix i do. 
 
In short its seting up your mix bus like this.
 
MY mixBass for example is  
EQ plugin
Mix Bus Compressor
EQ plugin
Tape emulation (vu meter) 
 
The trick now is that a VU meter is really sensitiv to low frequencies not to high. So if i start a mix i start with my kick drum volumen- It should peaki at -3dbvu (tape emulation VU Meter).
Things to know if you double an instrument the dbvu goes up by 3dbvu. So bass and kick have nearly the same frequencies. They are in the same ballpark. 
So my kick peaks around -3dbvu. If i bring the bass up -  bass+kick should peak at 0 dbvu.
Now build your mix around this gainstaging. And your bass should be allways in the ball park.
 
There was a youtube video horrible long but he explains it. I ll search it and post the link here.
 
EDIT
www.youtube.com/watch?v=lF2oNsiZWjc
There you go horrible long but worth looking expecially the bass volume trick ist woth this 50 minutes ;)

Thanks Smurf. I'll see what I can do with that formula. I've actually been thinking of VU meters due to some previous posts from Bitflipper. Didn't even think of the tape emulator having one. I was gonna go buy a VU plug-in. Great information! Thanks again.
 
~Jeff

Can you tell me what the difference is between the Voxengo and the Spectrum Analyzer in the Pro Channel EQ?

SPAN has far more options, such as slower averaging modes that are more helpful for mastering, the ability to freeze the display for taking screenshots and making comparisons, a correlation meter, clipping sensor, Mid/Side mode and a resizeable display. There are lots of spectrum analyzers out there, including some much more sophisticated than SPAN, but they're not free.

There has been a great deal of good advice, but there have also been a couple of "old wive's tales" that really need to be put to rest...
 
the decibBel is, as stated, 1/10 of a Bel, which was a unit-less measurement introduced by the Bell System to make transmission loss easier to work with. The original unit of measure (really testing my memory here) was Mile of Acceptable Cable. No, I'm not kidding!

None of which matters... what matters is that the dB is a ratio, and specifically, a power ratio. By definition it is an rms measurement - it can not be properly used to represent peak levels in analog measurements (more on this in a moment.)
 
rms means root-mean-squared, another handy little mathematical too that lets us calculate the effective power in a periodic electrical (or magnetic) wave. It is equal to the DC voltage that would produce the heat (dissipated power) in a pure resistance. All if which is important if you are a measurement geek, for us the things that matter are that it is used to measure the effective power in a periodic wave, and there is very little music that can be represented by a periodic wave, but all music is made up of periodic waves.
 
The dB ratio is equal to 10 times the log of the ratio of two power levels. If you know Ohm's Law you can manipulate this a little bit and come up with the forumla of dB = 20 times the log of the ratio of two amplitudes or levels (or in our case, voltages.) And for the curious, the forumula for the Bel is the log of the ratio of two power levels. But if you were curious you probably already guessed as much.

dB becomes much more useful when you replace the bottom half of the ratio with a known value.
 
In the beginning there was dBm0 which meant the ratio of the power in a circuit with respect to 1 mW across 600 ohms. This is still quite useful in telephony and RF, but it doesn't help us much, matched power transmission went out of style in the  early 1970s.

We are much more interested in dB scales that reference voltages. The first such scale used the equivalent voltage one would measure in that 1 mW signal across 600 Ohms, or approximately 0.7746Vrms. We refer to that today as 0 dBu, and there is no requirement for impedance, it is simply a voltage ratio.
 
It remains the basis of the "professional" audio interface. +4 dBu has become the reference level for a great deal of equipment, most often balanced inputs and outputs. That is not a requirement.

As consumer "Hi-Fi" gear grew in popularity the balanced interface was replaced with a single-ended interface, and the nominal operating level was changed from +4 dBu to -10 dBV. The dBV scale uses as its reference 1 volt. A lot easier to do the math, but it never really took off in professional circles. (Even today many broadcasters think +4 dBu is a silly standard, they prefer +8 dBu.)

Wandering off just a little bit - but there is a good reason - lets convert our nominal levels to their voltages:
 
+4 dBu = 1.227Vrms = 3.472Vp-p (and approximately 1.78 dBV)
-10 dBV = 0.316Vrms = 0.894Vp-p (and approximately -7.78dBu)
The two nominal levels are approximately 11 dB apart - just for grins!
 
OK, so what about this pesky digital audio?

All of the sudden we need to be able to think in terms of peak voltage if for no other reason that we do not wish to clip our converters. From this was born the dBFS scale, and it is slightly different because the reference is the maximum, there is no way - by definition - to go over 0 dBFS.

Which, as it turns out, really doesn't matter a whole lot either!
 
What matters, and what many folks have been trying to say, is that the relationship between 0 dBFS and the maximum analog voltage you expect to need is the important thing.

For example, lets say I have a microphone preamplifier that is capable of putting out +24 dBu (about 12.7Vrms or about 34 Vp-p). If we set 0 dBFS to equal +24 dBu then our nominal operating level will be about -20 dBFS if we are thinking in terms of +4 dBu analog audio.

That paragraph may trip up a lot of folks so I'm going to leave the math portion of this post for now - please feel free to offer questions, corrections, etc...

So when audio became a thing some bright folks noticed that neither a peak reading nor an RMS reading meter accurately predicted squat - you couldn't use it to manage modulation in your transmitter, and you couldn't use it as a benchmark about loudness. And that's what we care about!

So a new measurement was developed - the Volume Unit. You've seen (I hope) VU meters on audio gear. What makes a VU meter special is the ballistics, or how quickly it reacts to positive and negative going changes. It is WAY to slow to show a typical musical peak. But once you get accustomed to it you can get a pretty good sense of relative loudness.
In pro audio 0 VU = +4 dBu, in consumer gear 0 VU usually = -10 dBV. AHA, that dB stuff does matter eh?

No one that I am aware of has come up with a digital meter that mimics the ballistics of an analog meter. The Durrough meter plugin from Waves comes REALLY close. There are others too, I'm sure, but  I stopped looking a long time ago and just use analog meters<G>!
 
The meters in most software platforms do a really good job of reporting peak levels, and for the most part they do a decent job of reporting RMS levels. They just don't rise and fall with the same timing as the old mechanical meters.

Which may well be a good thing! In the  old days we didn't care much about peaks because there were about a dozen things in the audio path that would squish them anyway - especially as we got closer to the power supply voltage.
 
Not so in the digital realm - if  the analog circuit doesn't do too much damage the digital part will faithfully reproduce what ever we feed it - or create. So we really need peak responding meters in our software, we just  need to remember that they do not give us much of an idea of loudness. (A really short peak can reach the power supply rail and be effectively inaudible. How's that for trickery?)
 
Last thing for tonight (my finger are getting tired) - channels do not add algebraically unless the signals are identical land in phase. Say what?

Well if you've followed along this far you'd know that if I add two identical signals together I'll end up  with a signal that is 6 dB louder. But individual tracks are not identical (who really wants to listen to 16 tracks of 1 kHz Sine wave anyway?)

So they don't add up, and in fact can be more - or less - than the 6 dB difference one might expect.

By the way, using sine waves is an old trick and very handy - if I apply a -6 dBu sine wave to two tracks and then send them to a buss the buss will measure 0 dBu. It's a terrific trick for balancing a stereo circuit, or two monitors. And we used to use tapes with a single track across two head gaps to align tape machines using the same principle.

I know that's a lot, and I know there is a lot more, but I thought  this might be a good starting point.

Great advice from many here. I would only add that anyone mixing in less than ideal spaces, or who would like to test their mixes in various virtual environments should check out IK Multimedia ARC2. I have found it far more useful than I ever imagined it would be.
 
http://www.ikmultimedia.com/products/arc/