cumulative volume of tracks

bluesfordan

Member
I bought Logic Pro the other day and have been experimenting with it. Pretty much the same things I've been doing with GarageBand. The level of detail is light years beyond what I've been doing and that's not a bad thing. I'm glad that I've mucked around with GB for the last 2-3 years because otherwise I'd be intimidated as heck. But I've got the basics down and they're pretty much the same program.

A couple of observations. The first being that my humbucker guitar and single coil guitars using the same settings on the AI gain are surprisingly different. The humbucker was lower but more consistent, the single coils were louder but wider variances. I did three tracks, the first with the humbucker, the next 2 with the single coil.

with the improved metering, I could not only see when I exceeded +0 but by how much. What I found interesting was several times the guitar tracks were staying below 0+ individually but when they hit the same note at a certain point of a phrase, the stereo out bounced into the red.

through experimenting with the native plugins and adjusting the track faders (mostly tracks 2 and 3 with the single coil), I was able to get a good sounding mix that didn't hit the dread red zone. I was using the headphones since I have neither real monitors nor treated room to give the playback justice.

I don't know what the output of the single coils are but it was kind of counter intuitive that there would be so much more signal at the same input gain level. It didn't sound that much higher when I was recording, but I had to pull the faders down almost -9db to stop the saturation at the stereo out when I was playing it back.

Do you normally have to lower the gain input level for single coils if you're going to mix them with humbuckers? Is this what they mean by fixing it in the mix?
 
Not sure what pickups you are using, but if you are going direct to the interface with a cable I would be surprised if the track is hotter with single coils, at least in general. A single humbucker is 2 pickups, after all, so typically going to have more volts being generated, even with the phase/hum cancellation going on.

With no sims/fx in the channel, both set to the same input, record the single coil and then the humbucker, both guitars wide open. The channel faders should be at zero. You should have the interface gain set so the track peaks well below zero. Where are you seeing it go above zero? (If it is, you need to turn down the interface gain.)
 
And for those who don't want to read a link, basically: you should be tracking in the -18 to -12 dBFS range!
 
And for those who don't want to read a link, basically: you should be tracking in the -18 to -12 dBFS range!

Google wasn't much help in trying to find out what the meter in Logic Pro is using for a scale. It only says dB but is dBFS implied? Assuming it is, with my track fader at 0, I should be aiming for peaking to not exceed -12 dBFS, correct?
 
Google wasn't much help in trying to find out what the meter in Logic Pro is using for a scale. It only says dB but is dBFS implied? Assuming it is, with my track fader at 0, I should be aiming for peaking to not exceed -12 dBFS, correct?

If the top of the meter's scale is 0dB then it's dBFS. For sources that have sustain, you want the "average" level of the signal to be around -18dBFS. Obviously it's not going to stay on -18 so try to get it to cross -18 fairly regularly and let the peaks fall where they may as long as they don't get near 0dB. For impulsive sources like drums you can't really go by an average since they drop so fast from their peaks. In that case just try for peaks around -12dBFS. I do that but they usually creep upward to around -6. I don't worry as long as there's at least some headroom.
 
thank you all very much. I did a 6 track test last night, purposefully dense. The difference made by turning down the input gain far exceeded my expectations. It doesn't look that different in the tracking section but having the mixer section with its metering made setting the input infinitely easier.

Using only hardware compressor before the AI, with only one of the six tracks slightly hotter (-9 vs -mid teens), it kept the stereo out below -6, without any effects. no compression. no eq. And probably the best raw sound I've ever gotten in digital.

yeah, I'm a little slow on the uptake :D thanks for putting up with moi.
 
Just as a matter of interest and learning.

If on a sound console you (say) put a 1khz tone (just makes the explaination a bit easier) into a fader and raise its level to (say) 0db (on its Vu meter) and then raise the main output fader so that it also shows 0db on the main output Vu meter, then mute the input channel (do not touch its level), then do exactly the same for another input channel (but not touching the main output fader) it should also show 0db on the output Vu, now mute that input channel, repeat this same input channel process with a couple more channels.

Finally with all the input channels muted, un-mute the first input channel (will show 0db on the output meter), now leaving it un-muted, proceed to un-mute the second input channel and you should see that the level of the main output meter has risen by about 3db.

Now progressively un-mute all of the input channels and you will see that the output meter rises by about 3db each time an extra channel is unmuted.

What this means is that to maintain a 0db output, you will effectively have to lower the main output fader by about 3db every time you un-mute a channel.

However doing this to maintain the 0db output level is not necessarily good for the console's electronics, especially in the output section.

Therefore, it is necessary to adjust the gain of the inputs to accommodate this increase in voltage being applied to the output section's circuitry.

In live PA it is often considered an advantage to set the output fader first to a reasonable fader position (usually about 75% of its travel) and then basically not touch it, but mix the sound (balance) as required by only adjusting the input gains and input faders. With the main output fader being used largely as a fine tuning fader during the actual performance.

Although I have basically been referring to live sound (FOH) mixing, the same principle applies to sound recording and whether using an actual mixing console or a DAW.

Also, when I have used 0db as the reference, that could be any other set level (eg -15db in the digital world) but the same rise in level will be noted, and it does not have to be only tone, although this was easier in the example above and I am also assuming that the same tone and at the same level was applied simultaneously to every input channel in the above description

Hope this gives a bit of an idea as to the process and initial posting by "bluesfordan".

David
 
I've also noted something else. When I first got Logic Pro and was experimenting with it by recording multiple tracks one after the other, I noticed the stereo output level kept getting higher and higher even though I was keeping the input gain and track level down in the -18 to -12 range. Then I figured something out. I would create a new track but not disarm the input monitor of the previous tracks. I guess because the AI input wasn't changing the input signal was being monitored on each of the subsequent tracks, multiplying and ganging up on the stereo output. Once I disarmed the input monitor of all the tracks except the new one, I was getting a much better signal at the stereo output. Maybe this is just a quirk of Logic Pro, I don't know.
 
Actually, you get a 6dB rise every time you double channel count of identical signals. For different signals it's approximately 3dB per doubling.
 
Last edited:
Cubase can do the same thing if you aren't quite up on the pathways - the input channel is on the output, plus the track where it's going, and if you've got effects running, then they too add to the apparent volume. Mind you, so did my old analogue desks when you were tracking with groups to split the outputs to the record outputs.
 
Actually, you get a 6dB rise every time you double channel count of identical signals. For different signals it's approximately 3dB per doubling.

I've always sucked at this math, but doesn't that depend on how your meters are configured?

It's, what? 3 dB is double the voltage; 6 dB is double the physical power coming out of the speakers, and 10 dB is double the perceived volume, right?
 
I've always sucked at this math, but doesn't that depend on how your meters are configured?

It's, what? 3 dB is double the voltage; 6 dB is double the physical power coming out of the speakers, and 10 dB is double the perceived volume, right?

I believe it's the other way around from what you said, doubling voltage is +6dB while doubling power is +3dB. That's why doubling channel count for identical signal is +6dB. But in the real world you don't have a bunch of tracks of the same signal, you have signals that are non-correlated. They add in a more random way, so doubling track count to the master bus is approximated as +3dB. This also explains pan law, because in voltage terms perfect summing between left and right speakers would give you +6dB, but acoustic summing is never perfect so it's approximated to something like 3dB, with the center position being at -3dB compared to fully panned. Other common pan laws are 2.5dB and 4dB.

Audio meters are generally measuring voltage (except for dBFS, which is measuring a correlation of voltage, which is probably why 1 bit represents 6dB). I think many meters on amplifiers measure power.

Doubling the watts of an amplifier allows it to put out +3dBSPL because SPL is a power measurement. But you can't assume just switching out an amp for one with twice the wattage will give you +3dB because it may not have the internal gain to do it automatically. You might have to feed it with more input voltage. Boundary effect is also a power effect. Each time you add a boundary you get a 3dB boost at lower frequencies (below 100-300Hz, depending on circumstances).

I can't remember if 10dB (1 bel) of power or voltage is a 2:1 perceived difference. I suspect it's power.
 
ouldersound Guy,

Although it was a VERY long time ago, when at Uni we were taught audio in valve language, I am sure that I was taught that if you added two identical audio signals (ie voltage), that the level would rise by +3db and that doubling the output power on a loudspeaker was a +6db increase in volume (ie SPL).

I will have to either do a test on one of my old analogue consoles, that has analogue Vu metering or dig out my now VERY dusty Uni books to be sure.

Talking of being taught in valve language and to give you an idea of the era (!!!!!) --- or is that showing how old I am !!!!!

One night the lecturer entered the lecture theatre and proceeded to say "if anyone has anything that is super urgent that needs doing tonight, I will not be annoyed if you leave, as tonight's lecture will not be part of the exam, but I have to give the lecture all the same" to which he put his hand in his pocket and pulled out a transistor and showed it to the class stating "this is what is called a semi-conductor device, but do not be to concerned as it will never take off" !!!!!!!!!

David
 
Actually, you get a 6dB rise every time you double channel count of identical signals. For different signals it's approximately 3dB per doubling.

Yes and if we stick to voltages, two summed "correlated" sources such as sine tones of equal amplitude will result in a 6dB higher level (look up the basics of summing op amps)
The same goes for two identical music sources, though why one would use up an extra track for that I cannot say?

Uncorrelated sources such as resistor noise add as a 3dB voltage increase. If you simply mono a true stereo source, CO-I say you get a signal inbetween the two limits.

The remarkable result of all this is that you can make a MUCH quieter balanced line input using 8 op amp per ch than with one!

Dave.
 
Back
Top