Volume levels of everything

[HubertPolonski]

Hey everyone!

I'm pretty new to Reaper and music production in general, and I have a few questions about volume levels, everywhere.

1) I hear everything through a pair of DT 770 Pro 80 Ohms beyerdynamic headpohnes. They're plugged into a somewhat old sound interface, the TASCAM US-122 MKII Usb interface.

It has two knobs: a MON MIX knob, that sort of balances the audio between being 100% what's happening in reaper, and a 100% what's coming in as input. Then there's the Phones/Line out, that I guess is the volume control. Now, is there some trick to figuring out optimum settings on these? Do you know, if the MON MAX is all the way over to the computer, does that mean MON MIX is actually providing 100% or MORE? Is it amplifing?

As for the phones/Line out, how do I find out whether what I'm hearing in the headphones, matches with what comes out on someone elses computer (more or less).

2) In Reaper (and any daw), at the volume sliders for each item, we have these 6, 8, 30, 42, levels and the slider will say if it's 00db or +/-. Is there a value that the audio should NEVER reach? Any good rules of thumbs here?

3) Everything I'm recording is MIDI, with orchestral VSTs that use dxf/crossfaders or maybe Keyswitches (all PLAY). Do you know of any good resources where I can learn about the volume that sort of comes from the crossfaders (imagine the trumpet being at its weakest) to the audio levels of the item, to the audio levels of the speakers and so on.

Hope I'm making sense! You don't have to answer directly, any resources that touch upon these topics would be much appreciated also! I've tried googling, but think I'm having the keywords a bit wrong and I'm not really finding what I want to see.

Thanks!!

[DVDdoug]

if the MON MAX is all the way over to the computer, does that mean MON MIX is actually providing 100% or MORE? Is it amplifing?

I don't know if it's amplifying or not but if it's 100% computer, that means 0% direct. "Amplification" is kind-of meaningless because there is (uncalibrated) digital-to-analog conversion, plus a headphone amplifier with unknown gain (or attenuation) so a different interface might be louder or quieter with the same settings.

The purpose is to allow you to monitor yourself while you play/sing with no delay through the computer while also monitoring a backing track from the computer. That's a nice feature because not all interfaces have it and it can often be a pain getting latency (delay through the computer) down to an acceptable/unnoticeable amount.

As for the phones/Line out, how do I find out whether what I'm hearing in the headphones, matches with what comes out on someone elses computer (more or less).

As far as loudness? There is no way to know. Different computers, stereos, amplifiers, etc., put-out different voltage/power and speakers & headphones very in sensitivity and with speakers it also depends on how far you are from the speakers. Plus, everybody has a volume control. The only place loudness is calibrated is in movie theaters.

But, you can listen to a known-good commercial recording in the same genre and compare the loudness of your final product. Or, you can use an LUFS analyzer to check the digital loudness. (IMO - Scanning/analyzing the file as a whole is more useful than watching a loudness meter.)

2) In Reaper (and any daw), at the volume sliders for each item, we have these 6, 8, 30, 42, levels and the slider will say if it's 00db or +/-. Is there a value that the audio should NEVER reach? Any good rules of thumbs here?

Digital amplitude is measured in dBFS (decibels full-scale). 0dB is the "digital maximum" so digital dB levels are usually negative. It represents the "highest you can count" with 8, 16, or 24-bits. (Everything is scaled when you record and play back so a 0dB 24-bit file is not louder than an 8-bit file.)

Analog-to-digital converters (recording, digital-to-analog converters (playback),"regular" WAV files, and CDs are all integer-based and they are hard-limited to 0dB. If you try to go over you'll get clipping (distorted flat-topped waves).

Now... REAPER uses floating-point numbers internally so there is virtually no upper limit and REAPER itself won't clip. So, you need to avoid clipping while recording (if you record analog), you should "watch your master levels" while monitoring to avoid clipping, and make sure your final render is not clipping. Otherwise, your levels are not critical. One technique is to render-to floating-point WAV (which can go over 0dB). Then, re-import and normalize to 0dB (or some people like to normalize to -1dB or so).

Peak levels don't correlate well with perceived loudness. Perceived loudness is more-related to the short-term average (or RMS) and the frequency content. So if you normalize/maximize all of your files for 0dB peaks they won't sound equally loud.

It's unlikely that you'll get the same loudness as a professional mastering engineer can get without excessive "damage", and you may not want to use that much dynamic compression. (Dynamic compression and limiting can bring-up the average level without boosting/clipping the peaks.)


...Acoustic loudness is measured in dB SPL (sound pressure level). The 0dB SPL reference is (approximately) the quietest sound that can be heard, so SPL levels are positive.

Like I said, except for movie theaters there is no standard calibration between digital DB levels and SPL levels, but there is a direct correlation. If you reduce the digital level by 6dB the acoustic level will go down by 6dB.

And, you can calibrate your studio (although it's not so easy with headphones because your SPL meter isn't accurate with headphones). For example - K-system Monitoring.

[ashcat_lt]

1) What's coming out of the computer to the interface itself has no inherent analog volume. It's basically just a number representing some fraction of the system maximum. Essentially it's somewhere between -1 and 1, where 1 is exactly as "loud" as the analog level can ever get. What that is in absolute terms depends on the interface itself. The Mix knob all the way to computer means you will only hear what's coming from the computer, but the question of whether it is "amplified" is pretty much moot.

There may be (often is) a difference between how much voltage goes in the input and how much comes out the output even with everything at "unity" in the DAW itself. That depends on the design of the interface and sometimes on which hole you're plugging the input into and sometimes settings like -10/+4 switch and of course any gain knobs along the way.

Most interfaces (even cheap onboard sound) is actually pretty good nowadays and generally whatever you hear on yours will be about the same as far as frequency response, dynamic range, and everything that matters except for overall volume. There is nothing you can do to make sure that somebody else's machine is going to output the same p2p voltage as yours. It'll sound as close to the same as their speakers will get, but the volume is up to them.

2) That maximum output that we were talking about - the unit 1 that we defined - is shown as 0dbFS on Reaper's meters. The interface can't get louder than that, so if you try, it goes as far as it can and "clips off" anything above. It sounds like distortion, but actually how that distortion itself sounds depends some on the converter that's trying to make it analog. This is where your listeners might have a significantly different experience from yours. If there's a lot of actual 0db clipping in your file, it will be harder to predict what others will hear. So keep your Master levels below 0dbFS. If you like the sound of the distortion from clipping, you can add that with plugins, but at some level below 0, and then actually know that the distortion will sound about the same everywhere.

3) I'm afraid I don't understand this question.