Clipping without reaching 0dB

[7heodoros]

Hi,

I am using Reaper and today I noticed something weird.
I am recording my voice as usual and I feel like it's clipping, but my input level on Reaper shows like -9dB.

My audio interface has an LED panel where it shows that it's clipping. It's definitely distorted. I also notice it by looking at the waveform.

Why does Reaper show -9dB? Is it just some settings or hardware (broken interface)?

I know I can just turn the gain nob down, but that doesn't solve the problem. The problem is that the interface and my DAW are out of sync. My interface shows that it's clipping (and it actually is clipping) while Reaper says I still have a lot of headroom.

See attached images.

Any ideas?
Thanks in advance for your help.

[Philbo King]

Open s new project. Turn down your speakers.
Put the tone generator on a track and set it for 400 Hz at -18 dB.
Make a small loop and hit play.

Set your interface output level so it just barely hits 0 dB.

Loop that output back into a line input with a cable. Set the input level so the input meter barely touches 0 dB.

Arm a separate track and look at the level coming in from the loopback input.

Set the interface control so the meter reads -18 dB.

The idea:
- Anything inside your DAW uses dBFS (decibels relative to digital full scale).
- Anything at your interface uses dBu or dBv (decibels relative to 1 volt).
- The defacto standard to translate between these two is 0 dBv = -18 dBFS.
This allows you 18 dB of headroom before something clips on the way in or out.

Depending on your audio interface and it's capabilities, a different calibration level might work better. But it's a starting point.

[DVDdoug]

My audio interface has an LED panel where it shows that it's clipping. It's definitely distorted. I also notice it by looking at the waveform.

[B]Trust the interface. It's the analog-to-digital converter inside the interface that (normally*) clips at exactly 0dB.

Why does Reaper show -9dB? Is it just some settings or hardware (broken interface)?

That I don't know. Something is "misadjusted"

- Anything at your interface uses dBu or dBv (decibels relative to 1 volt).

No. The meters should be calibrated to dBFS to match the digital level.



* It's possible for something else to clip. For example, an external preamp can clip but with a preamp built-into the interface, the preamp always has enough headroom to clip the ADC. But if you get clipping during recording it's almost always the ADC clipping at exactly 0dBFS.

[Philbo King]

My bad. You're right, Doug. I got so used to using the meters on my mix console, I forgot. I haven't looked at my interface meters in several years.

[7heodoros]

Originally Posted by Philbo King

Open s new project. Turn down your speakers.
Put the tone generator on a track and set it for 400 Hz at -18 dB.
Make a small loop and hit play.

Set your interface output level so it just barely hits 0 dB.

Loop that output back into a line input with a cable. Set the input level so the input meter barely touches 0 dB.

Arm a separate track and look at the level coming in from the loopback input.

Set the interface control so the meter reads -18 dB.

The idea:
- Anything inside your DAW uses dBFS (decibels relative to digital full scale).
- Anything at your interface uses dBu or dBv (decibels relative to 1 volt).
- The defacto standard to translate between these two is 0 dBv = -18 dBFS.
This allows you 18 dB of headroom before something clips on the way in or out.

Depending on your audio interface and it's capabilities, a different calibration level might work better. But it's a starting point.

Thanks a lot guys for your help.
I am new to calibrating interfaces. I am using a Presonus studio 2|6 and I am trying to figure out how to loop the signal. I only have 2 XLR inputs and 1 midi. The outputs are 1/4". I don't have an XLR to 1/4" adapter or cable. Is there a way to loop the signal differently? Sorry for the noob question but as I said, I never did this before.

The driver has a loopback option. Would a virtual loopback do the same thing? (see image)

[7heodoros]

Originally Posted by Philbo King

Open s new project. Turn down your speakers.
Put the tone generator on a track and set it for 400 Hz at -18 dB.
Make a small loop and hit play.

Set your interface output level so it just barely hits 0 dB.

Loop that output back into a line input with a cable. Set the input level so the input meter barely touches 0 dB.

Arm a separate track and look at the level coming in from the loopback input.

Set the interface control so the meter reads -18 dB.

The idea:
- Anything inside your DAW uses dBFS (decibels relative to digital full scale).
- Anything at your interface uses dBu or dBv (decibels relative to 1 volt).
- The defacto standard to translate between these two is 0 dBv = -18 dBFS.
This allows you 18 dB of headroom before something clips on the way in or out.

Depending on your audio interface and it's capabilities, a different calibration level might work better. But it's a starting point.

I was trying that with the virtual loopback and even though I wasn't able to do exactly what you told me, it helped me to find out that the problem disappears when I use my ASIO Driver instead of Direct sound or Wave Out. Any idea why? (image)

[domzy]

Originally Posted by 7heodoros

I was trying that with the virtual loopback and even though I wasn't able to do exactly what you told me, it helped me to find out that the problem disappears when I use my ASIO Driver instead of Direct sound or Wave Out. Any idea why? (image)

There are volume controls for input & output in Windows audio properties - these will have an effect if you are using windows generic drivers

[ashcat_lt]

I don’t want to confuse the OP issue here, but I think that’s sorted, so I’m gonna go off on this little tangent about loopback and “calibration” that really has nothing to do with what they were seeing, but is a thing people might run into.

The thing is that many interfaces, including the most common ones from big manufacturers like Presonus, Focusrite, etc, are NOT matched from input to output. Their maximum input level is different from their maximum output level in almost every case. Now you’d kind of expect a mic input to have some gain over the corresponding line input (on my Tascam it’s a minimum of 6db hotter) and most instrument inputs are 9-10db louder than the line in, but one would sort of expect the line ins to be essentially unity gain through to the output so that, say, +4dbu input would be the same +4dbu at the output as long as everything along the way is set to 0db gain. That will almost never be true, though. Very often the output will be significantly louder. I’ve seen interface specs where the maximum input level is like +8dbu while the output goes to like +18dbu or more. Those both correspond to 0dbFS, which means there’s 10db or gain when you thought you had everything at unity!

This only really matters if it matters, like if you’re trying to do hybrid mixing with hardware as inserts or other things like that. Reamping is an example. You’d kind of like for the signal going back out to the amp to be the same it would see from the guitar, but even if you use a good active buffer and record through the line in, it’s going to pound the amp unless you compensate somewhere along the way. That’s not so bad IF your manufacturer actually provides the appropriate specs, but with some it’s pretty tough to find that information.