Hi,in a tutorial the guy says make sure bass level is AVERAGING about -20 dB.
in cubase 14 pro on channel window how can i check average level?
On the right zone meter(master tab) is it the RMS Ma?

 

Open the mixer and see if the faders are hitting -20dbfs. This is only a guideline, though, some people will go as high as -12dbfs as long as the signal is not clipping.

Edit: to be clear, the fader metering, not the fader position.

 

That must be -20dB RMS, could peak up to about -12dBfs depending on how much you compress it. It's a good value so you never have to worry about the headroom and mix in peace.

 

Yeah, you're on the right track - in Cubase Pro 14, the RMS value in the right zone meter (Master tab) shows the average level. If it says "RMS Max" or "RMS," that’s what you want to look at for the average loudness over time.

 

Actually there is no fixed value on the input volume: the digital recording volume has its limit in the bitrate used by the audio interface: 16bit has 96db max, 24bit has 144db. Depending on the bitrate, a different clipping value corresponds.
Technology is running: at the beginning we recorded in 12bit, then in 16bit and in 24bit: we now have the 32bit floating point technology coming and this means having 1528db on input: this value exceeds any real-world sound pressure variations. In simple words, this is the end of clipping during recording.
At the moment, 32bit is mainly used on movie sets, when there can be drastic variations on the sound pressure (imagine a dialog and then a jet passing by). From what I know, there are a couple of brands that already produce field recording devices in native 32bit: Zoom and Tascam. There is an audio interface from Zoom, dedicated specifically to the 32bit recording studio, but I have not had the pleasure of testing it.
The future is obviously in 32bit floating point
https://www.zoom-europe.com/en/audio-interfaces/zoom-uac-232-c

 

Last edited: Jun 8, 2025 at 9:34 PM

There is no "formula" for this - it's way too subjective. Does it sound balanced? If not, then you need to act accordingly, and not adhere to some kind of "method" for setting levels. If you do, it's guaranteed your mix will come out sounding messy.

 

I have had one of these for a couple of years now. They are little audio recording powerhouses. I walk around with that audio pack, cans, and a mic and I hardly even know I'm carrying anything. I have used it for field audio for video (it does SMPTE directly) and for audio sample gathering. The audio inputs on those are clean enough to use in studio as a recording interface and you are correct - it will record up to 96Khz in full 32bit. I have never gotten back and had a recording that was unusable from that device.

 

This it bit depth, bit rate is SF X bit depth X number of channels, it measures a data flow.

Converters clip when the voltage they measure exceeds what they can handle. So, a converter clips at the same level either in 24 or 16 or 12 bit. What better bit depth brings is a lower noise floor and the ability to record avoiding clipping by settting a lower average level. Bigger Bit depth dosent add headroom but what could be called 'footroom'. It doesnt change anything about the highest recordable level, it just reproduces lower levels.

Unfortunately, Zoom doesnt give any technical details about its 32FP ADC. I'm neither electronics engineer nor converter designer, but I think that a floating point converter is impossible.
As clipping occurs at a fixed voltage in the analog side of the ADC, I dont see how a floating point converting stage could work.

In 32 bit float, the audio signal is 24 bit and the 8 remaining bit are a mantissa which allows the 24 bit signal to exceed its theorical range.
I think that the Zoom 32FP ADC is in fact a 32 bit fixed point ADC with an automatic input gain and further floating point processing. 32 bit fixed point has 192 dB dynamic range, a lot more than any analog input stage. It may automatically adjust at a relatively low gain, ensuring that no clipping will happen and then recode this signal (where the max level is far below the clipping point) into 32 bit float audio data.

As I said, I'm no specialist, what I think may be wrong and I'd love to see some people who actually knows to chime in.

 

It totally depends on the mix, genre, amout of elements, etc. There is no "mix by numbers". Simple gain-staging along with good mixing will determine the "correct" level of bass.

 

Yes, maybe I didn't express myself in a very technical way but we're talking about the same thing: 32bit floating point is the end of clipping. I've never used one of these new audio interfaces and I can't testify if they work or not, but I have confidence in technology and I think we'll get there anyway