The first one is: what about 16-bit? Don't I need to push my levels way up in order to minimize noise?
Answer: Yes, unfortunately. But try to be reasonable. Boost the levels until there is no clipping but the signal is healthy enough that you can't hear low level noise. Compressing on the input before the recording interface can really help with this. If you know how to use a compressor and are still forced to record at 16-bit, I would say it's a necessity to compress the signal before committing it to the hard disk. If you record at 24-bit, it's not nearly as much of an issue.
The Digital meter to the left has a hard upper limit of 0dB. Anything above that limit will register as awful, loud, crunching, grating noise because there are no bits above the first bit. Make sense? If not, read this post again, then this one
This stuff can be really dry and boring for people, but I think important to understand what's happening behind the scenes so that you know why you need to make certain decisions later on.
Recording level means different things depending on what kind of recording we are talking about.
Analogue recording uses a different scale to measure peak level than digital recording. Confused? Don't worry.
In analogue gear, the level of signal can be measured with something called a VU meter like this one ... (if you remember from a previous post, audio signal going through cables is just very low voltage electrical impulses) ...
With the VU Meter above, going above 0dB will not produce that hard noise, because 0dB is not a hard limit in analogue gear (in fact, it's the sweet spot). "Wait, wait that makes no sense!" you say. The key to unlocking the mystery is that the digital DB scale and the analogue dB scales are completely different scales! The digital scale is annotated dBFS, while the analogue scale is annotated dBu or dBvu. So any time you see dBu, you are dealing with analogue audio levels, and any time you see dBFS, you are dealing with digital audio levels. Also, most FS meters are peak meters, while the dBu scale is used for judging average analogue levels (and it needs to be calibrated to your gear! VU Meter calibration is VITAL).
This is a completely different Beast than a Digital Full Scale Meter.
Did you know: almost every piece of analoge gear has a different absolute upper limit!!? For example, in my Focusrite 2i2, the clip level of the analog microphone inputs is +4dbu, but the clip level for the analogue line inputs is +20dBu! And the clip level for the analogue outputs is +10dBu. How confusing is that?!?!
This means that a signal from a microphone that registers more than 4dBu above 0dBu is going to hard clip the microphone preamp. Analogue Hard clipping just means that significant distortion will be added to the sound, sort of like a digital clipping sound but just a hair more pleasing to the ear. As a side note, it is possible to soft clip analogue gear which gives harmonics and saturation. This can be pleasing to the ear, but you want to be careful soft clipping your monitor outputs because it will smear the frequency spectrum!!
In my Focusrite 2i2, the outputs have more headroom than the inputs. Headroom is a term to describe the amount a signal can be turned up before clipping (it works for both digital and analogue). So in Focusrite 2i2, a digital signal of 0dBFS translates into an analogue output signal of +10dbu
So, here is where the rubber meets the road. The absolute best level to record your music at (assuming 24-bit), is line level, which is always 0dbu.
On the mic input, we have 4dbu of headroom before clipping. This means that going in through a microphone, +4dbu will translate to 0dBFS. We want to stay WELL below that level. At least 4dBu below so as not to clip the preamp, but at least 10dBu below so as not to clip the outputs.
The best bet? Record at the lowest acceptable volume for the input and output stage. Let's start by using the outputs as our guide for recording something with a microphone, since doing so will keep us well under the clip range of the microphone preamps.
Here's the math:
if +10dBu = 0dBFS, then
0dBu = -10dBFS.
Just subtract the maximum analogue level from both sides of the equation and voila, you have a good peak level to watch on the digital audio scale as you record. This formula works with any recording interface by the way as long as you know what the maximum dBu level is!
What if you are recording something on the 2i2 using the line input? Well, since the line input has way more headroom, the math changes a bit (but not much). With the line level input, +20dBu = 0dBFS. so ....
if +20dBu = 0dBFS, then
0dBu = -20dBFS
To keep things simple, I record, mix, and master at average level of below -15dBFS. I use an RMS meter to judge this, but you can also just watch the digital peak levels bounce in a DAW and eyeball what the average level might be. This means that as I add more elements to a mix and the levels start to push above -15dBFS, I either select all the faders and pull them all down in unison until I'm back in the sweet spot. Or I just pull the master fader down.
Why is it important to record this way? The reason is simple: because 0dBu is the sweet spot for all analogue gear, including the analogue inputs and outputs of every digital recording interface ever made!! It's also the sweet spot for MANY plugins!!
Its just that figuring out what 0dBu means in terms of the Digital scale can be tricky, but hopefully now you have a better understanding of how to do that and what's going on behind the scenes. And just like gear, plugins have their own maximum levels, sometimes expressed as dBu. It's insane, but it's worth figuring this stuff out so that you can keep as clean a signal path throughout your recording as possible. The results on each track can be subtle, but across an entire mix it makes a huge difference.
Here's to your recordings!
Ryan
Disclaimer: I do not work for focusrite nor am I advertising for them. I just like the gear of theirs that I've tried. It's great bang for your buck and is very high quality IMHO.
All Images via Wikipedia
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