USB Output level too low

[safjuan]

Hi

I'm using the helix as an interface going into Live 10.  I've tried both with the Line6 Asio drivers, and Asio4all, but the input level is really low.

I've checked guitar pad (it's off) and the big volume knob is set all the way.

 

Do I have to go into each individual patch and increase the volume, or is there a setting that I can't find that will raise the USB 1/2 output level across the board?

 

I looked everywhere in the manual and couldn't find anything

 

Thanks for your help

[rd2rk]

The Input Level should be around -18 to -12db. That is intentional, as it leaves headroom for post processing.

[OmniFace]

How low is it?

As noted above, you often want the average volume to be around -18dB in your DAW.  This is "dB Full Scale" or dBFS.  That correlates to around 0dB that you see on an analog VU Meter.   (0dB = -20 dBFS).

 

If you're too far below that, it means that your signal chain within the Helix is probably too low.  You'll need to turn up the Channel volume on each.  Or use a Gain block at the end of the signal.

 

If you're close, I wouldn't worry about it.  You'll only lose a little bit of signal to noise ratio, but guitar amps are already noisy.  So just turn it up in your DAW with the clip gain or a gain plugin to compensate if needed.

 

 

[safjuan]

I'll double check exactly how low it is.  This is with factory patches anyhow.  I was just surprised that there isn't a "USB output level" setting somewhere to raise volume across the board.

 

Without exact measurements, the waveform created by ableton is about 50% before it hits peak.  I can always just raise the level of that track, just worried about introducing noise to the recording by doing so.

 

I'll measure and report back.

[phil_m]

26 minutes ago, safjuan said:

I'll double check exactly how low it is.  This is with factory patches anyhow.  I was just surprised that there isn't a "USB output level" setting somewhere to raise volume across the board.

 

Without exact measurements, the waveform created by ableton is about 50% before it hits peak.  I can always just raise the level of that track, just worried about introducing noise to the recording by doing so.

 

I'll measure and report back.

 

It really wouldn't make any difference if you were able to raise the level with some USB output parameter prior to it being in your DAW or doing it in your DAW, assuming you're talking about just normalizing up to a certain level. Both things are being done in the digital realm, so the signal to noise ratio won't be affected.

This is another reason why recording via the USB out is superior to using the analog outs into a separate interface. These signal to noise considerations that come into play with the converters go away.

[zolko60]

The processed output level in -20 LUFS ballpark is OK and the direct (USB 7 usually) at -30 LUFS is also correct.

19 minutes ago, safjuan said:

Without exact measurements, the waveform created by ableton is about 50% before it hits peak.

50% may be at -6dB. It is too loud for -20LUFS and distorted tones. It should not exceed -10dB peaks unless you constantly monitor levels against clipping.
Such a packing under the ceiling is passe in XXI century unless you are soldier on mastering loudness war.

[zolko60]

4 minutes ago, phil_m said:

Both things are being done in the digital realm, so the signal to noise ratio won't be affected.

This is another reason why recording via the USB out is superior to using the analog outs into a separate interface. These signal to noise considerations that come into play with the converters go away.

Not true. SN ratio is affected because noise is recorded and remain constant. If you leave 10dB of peak headroom SN ratio of AD converted signal will be lower by 10dB cause it is constant.

[phil_m]

5 minutes ago, zolko60 said:

Not true. SN ratio is affected because noise is recorded and remain constant. If you leave 10dB of peak headroom SN ratio of AD converted signal will be lower by 10dB cause it is constant.

 

The signal to noise ratio of the original recorded track will remain constant. Normalization is simply a mathematical operation. The recorded signal doesn't change.

 

If you have noise in the original track and gain it up, of course, the final mixed track could have more noise. But if you're talking about raising the level of a digital track after the fact, it doesn't matter is it's done inside the digital source or in the DAW.

[zolko60]

1 minute ago, phil_m said:

he signal to noise ratio of the original recorded will remain constant. Normalization is simply a mathematical operation. The recorded signal doesn't change.

Oh really? During that "mathematical operation" noise is added because it masks aliasing noise which is worse.
Albeit this is not a reason normalisation makes any sense working in 24 bit integer of 32 floating point enviroment. The reason is some guys are familiar with "under a celiling" recording. I was too and changed my mind. Leaving 10-12dB of peak headroom is much safer and devoting some SN ratio is worth it.

[OmniFace]

On 12/30/2020 at 2:30 PM, zolko60 said:

Oh really? During that "mathematical operation" noise is added because it masks aliasing noise which is worse.
Albeit this is not a reason normalisation makes any sense working in 24 bit integer of 32 floating point enviroment. The reason is some guys are familiar with "under a celiling" recording. I was too and changed my mind. Leaving 10-12dB of peak headroom is much safer and devoting some SN ratio is worth it.

 

 

"noise is added because it masks aliasing noise"

I believe what you're referring to is Dithering, which is used when converting an audio signal into a lower bit depth. 

 

Normalization is simply raising or lowering the amplitude of the entire track.  It doesn't alter the sample rate or bit depth of the signal.  It merely says that a audio data point sitting at say -6 dB is now -4.7dB.  But it does that for all data points, meaning that means that the S/N ratio stays constant.  Both get louder or quieter in equal amounts at the same time.

 

However, using an external interface rather than the USB does mean the signal has to be converted from D to A, then A back to D again.  That said, I'd wager that converters these days are such high quality you may have to run them through conversion hundreds or perhaps thousands of times to hear a difference. 

Plus it means going through the interface hardware coloring the sound in whatever way that device does.  That can impact the S/N ratio because the interface will introduce its own noise, and setting the Helix output and hardware input will be important.

[zolko60]

On 2/5/2021 at 9:25 PM, OmniFace said:

Normalization is simply raising or lowering the amplitude of the entire track.  It doesn't alter the sample rate or bit depth of the signal.  It merely says that a audio data point sitting at say -6 dB is now -4.7dB.  But it does that for all data points, meaning that means that the S/N ratio stays constant.  Both get louder or quieter in equal amounts at the same time.

Oh, really? If you have three intiger numbers: 1, 12, 65 and you want to rise them proportionally as so that 65 is 100, then you have to mutiply them by 100/65, right?
1.538, 18.462, 100,

then to round them to integer values 2, 18, 100 (becacuse the sample rate does not change, right?)
What happens to those differences? Don't they form noise?

S/N ratio is the proportion of the loudest signal to the noise floor level. If the A/D converter has eg 105dB S/N ratio and you keep 10dB of headroom so your recorded signal can only have 95dB of S/N ratio, right? Still if it has 50dB S/N ratio before it hits A/D converter, it makes a small difference.

On 2/5/2021 at 9:25 PM, OmniFace said:

That said, I'd wager that converters these days are such high quality you may have to run them through conversion hundreds or perhaps thousands of times to hear a difference. 

You can actually make an experiment of putting the Helix fx loop before a high gain amp model.
 

[soundog]

For more flexibility over your levels, run your Helix output into an audio interface, You can boost it there to your heart's content. I run my geetars through an active DI that splits the guitar signal so I can record a dry and wet track (or stereo wet) simultaneously if needed. I stopped using the Stomp's USB for recording. I felt that the input level seemed a bit wimpy (technical term for low).

[OmniFace]

On 2/19/2021 at 11:23 AM, zolko60 said:

Oh, really? If you have three intiger numbers: 1, 12, 65 and you want to rise them proportionally as so that 65 is 100, then you have to mutiply them by 100/65, right?
1.538, 18.462, 100,

then to round them to integer values 2, 18, 100 (becacuse the sample rate does not change, right?)
What happens to those differences? Don't they form noise?

True, but 24-bits works with 16,777,215 points of amplitude.  So, while your example on small integers is correct that some noticeable rounding errors would occur, the scale you chose makes your example meaningless.  The number of audio points in 24 bit audio means the rounding errors are insignificantly small.

Assuming the audio is integer based, we'll divide this range up into 100 dBs for ease of use.  Here are 3 random points using -100db = 0 and 0db = 16,777,215 on the y axis

 

-9dB = 15,267,252.65

-42dB = 9,730,784.7

-86dB = 2,348,810.1

 

If we round them all we have at max difference of 1 in the 1's place.

 

-9dB = 15,267,253

-42dB = 9,730,785

-86dB = 2,348,810

 

We have more than enough integers to account for the shift.  If we compare the unrounded vs the rounded we have a difference of just a few 100 millionths.

 

-9dB =  1.0000000229

-42dB = 1.0000000308

-86dB = 1.0000000426

So yes, the signal is not technically identical, but the difference certainly negligible.  Even if this increased noise it would be unnoticeable in my opinion.