Two Mixing Questions for a Newbie

One more question: say you want all songs on a record to be the same level so the listener doesn't have to vary the volume song to song. How is this done?

One of the things I agree with Synthzizer on is: use your ears. Bring all your raw mixes into one project and make them all sound the same volume. There's no single trick for this, but I typically start with Clip Gain (as it's called in Pro Tools) and match all the levels by ear, generally keeping them around -18dBFS as a rough guide. Then I use a mastering limiter on the main bus to bring the levels up to my target range. In between those two steps I may do all sorts of processing depending on the specific needs of the raw mixes. Some processes may be on individual tracks, others may be on the whole project.
 
One of the things I agree with Synthzizer on is: use your ears. Bring all your raw mixes into one project and make them all sound the same volume. There's no single trick for this, but I typically start with Clip Gain (as it's called in Pro Tools) and match all the levels by ear, generally keeping them around -18dBFS as a rough guide. Then I use a mastering limiter on the main bus to bring the levels up to my target range. In between those two steps I may do all sorts of processing depending on the specific needs of the raw mixes. Some processes may be on individual tracks, others may be on the whole project.

Okay, thanks. When you say import the raw mixes are these mixed down to .wav files, or do you mean import each track from each song?

What is this process called? I can research it on my own if I know what it's called.

Also, at the end when you have say 10 songs for a record, how do you then put spaces/silence between them? Sorry for all the questions...I'm about to embark on a pretty big project recording years of my tunes and want to think about the end now so I don't run into trouble down the road.
 
Okay, thanks. When you say import the raw mixes are these mixed down to .wav files, or do you mean import each track from each song?

I mean import the stereo wave files of the mixdowns.

What is this process called? I can research it on my own if I know what it's called.

Most people would call this mastering, but really it's just audio finalizing.

Also, at the end when you have say 10 songs for a record, how do you then put spaces/silence between them?

Technically speaking, this is mastering. You have to decide what you're mastering for. Is it vinyl, CD, the internet? Each of these has specific requirements. There's specific software for CD mastering, though some DAWs can do it. With a third party plugin Reaper can produce a DDP 2.0 file ready for upload to a replication company, but it isn't exactly user friendly. For a serious release most people will have their project professionally mastered, which is generally considered to include the audio finalizing part of the process.
 
The Mix Bus of the Daw is next to useless for summing.
Daw faders are shite too.
Daws are next to useless for summing.
Now you need computer and programming degrees to mix in the Daw but then this comes out as square wave washing machine music anyway. Good luck.

Is any of this true?
I'm not going to lie. You sound like someone who's struggling with the transition.
 
Is any of this true?
I'm not going to lie. You sound like someone who's struggling with the transition.

You tell me? A few people would agree that a mixes track gain structure in the Daw if set up wrong causes the mix bus to be next to useless. Like I said.... So if one where to follow the advice of the soundonsound article I referenced to then this would help immensely.
The transition? No... Been there done that... Problem is everything must work in synergy with a digital Daw set up, including every plugin & O.S. This is a difficult proposition for the home recordist because alot of them are obsessed with buying new plugins that they think will be the ticket to fix there problems which they are not. Then having to deal with each company's licence system can prove to be a problem. This results in another bottle neck of system stability and errors abound with many hours spent on Forums for advice. The same is true for interfaces that dont perform or Daws that are unstable. Daws are tested in perfect conditions at there company headquarters but in the hands of a home recording computer is very different.
 
I think you are over-emphasizing "problems and difficulties" of working ITB.
The problem is not the DAW or the plugs or the computers....the problem is that many DAW users are also first-time home recordists, and I would bet that given an all-analog/OTB setup, those same people would struggle as much, maybe even more, because analog/OTB certainly requires even more precise/focused operation than ITB...even if it is a bit more forgiving on the headroom overs.
IOW...you can crap out things when summing OTB just like you can ITB if you don't know what you are doing with gain/levels.
 
A few people would agree that a mixes track gain structure in the Daw if set up wrong causes the mix bus to be next to useless. Like I said.... So if one where to follow the advice of the soundonsound article I referenced to then this would help immensely.
The transition? No... Been there done that... Problem is everything must work in synergy with a digital Daw set up, including every plugin & O.S. This is a difficult proposition for the home recordist because alot of them are obsessed with buying new plugins that they think will be the ticket to fix there problems which they are not. Then having to deal with each company's licence system can prove to be a problem. This results in another bottle neck of system stability and errors abound with many hours spent on Forums for advice. The same is true for interfaces that dont perform or Daws that are unstable. Daws are tested in perfect conditions at there company headquarters but in the hands of a home recording computer is very different.

Not being funny, but that's an entirely different set of issues.
If you choose your parts carefully, though, there's no reason you can't have a stable setup. Plenty of people manage to do that.
I understand computers are familiar to some and less so to others, though. It's easy to wind up with less than stable gear.

Now, if you'd said "Buying the wrong gear and not using it properly is shite." I might have agreed. ;)

You tell me?
Ok. I disagree with all the things I quoted above.
 
It can maybe help more if instead of volume you think of it in terms of electric charge, or put even simpler, plainly as "Energy". You can find out what input and output dBu your converter can operate at max before clipping. Here it is recommended that you use a converter that can input and output at least +24dBu, use as much balanced connections as possible at as high input dBu as possible (or slightly lower when noticing that you go beyond the amp's sweet spot) to utilize that headroom when you record. Once the signal is inside the DAW you can then trim the level to fit the available output headroom of +24dBu, depending on how many tracks you have. dBu is a unit that essentially converts to the volts (RMS) electric potential, which is kind of a little confusing. Why it is "potential" is because the electrical charge is relative to the Ohm. The electrical charge Ampere has Ohm according to: V = A * Ohm. Important also is that the scale is cumulative, so the difference in Energy utilization between let's say A) +18dBu and +24dBu vs B) +12dBu and +18dBu, is A) 6 volts (RMS) vs B) 3 volts (RMS). For this reason it makes a big difference if you use a +24+ dBu converter or say a lower one typically at +18 dBu. A high energy music platform kind of translates into high musical energy flow, you kind of want your music to have as high such impact as possible. Therefore, use a lot of hardware, push it a lot and let the power do the job for you towards a beautiful sound. Avoid getting too focused on the dBFS thing, it is after all only relative to the max electrical potential in your particular DAW (read audio interface/converter). Choose a converter that can operate at as high dBu as possible, which provides you a stronger overall signal, much better stereo image and more emotional music. A technique that can be handy to know about is that the stereo element separation is relative to the Volts RMS difference between tracks on each stereo channel individually (L or R). Therefore, cut down on the number of sound sources/tracks playing at the same time on each speaker, set the right RMS difference between those elements with volume faders and pan knobs and at the same time utilize as much dBu headroom as possible. You can further optimize it by distributing the frequency range on each speaker as efficiently as possible. This essentially creates a great and powerful stereo image. It is important to have a great headroom in your DAW. Mix Headroom = Mix Max RMS * Mix Max Dynamic Level, when roughly simplified down to practical/useful terms. You can try to use two stage compression instead of single stage. That you can achieve by A) sending the signal in parallel to compressor X, and then B) sending the result of A) including the original source to compressor Y. Exactly how this routing is done you can read about in Tape Op, online for free, by doing a search on "two stage compression". This can work well on sound sources that consume quite a lot of mix signal, for instance on drums.
 
Last edited:
Back
Top