Interface gain settings

[CaptOblivious]

Have been stuck in a rabbit hole for a while now after seeing numerous YT vids about setting input gain for software/plugins - many are saying that you should leave your A/I input gain at minimum and adjust plugin/software gain slider to achieve the tone.

I originally set the A/I input gain by wanking on open E chord as hard as I can and adjust the gain so no red, only spiking into yellow. I found setting that way everything was super saturated, could not get a decent clean tone and had to back way off on the volume slider to not clip. Setting the A/I input gain to minimum and adjusting in the software/plugin improved things, but YouTubers say you need to know your interface specs to adjust more accurately.

Well, maybe they can but . . .

I use an AXE I/O One with the following specifications for Instrument:

Input level, min gain: 0 dBFS is obtained with a +10.5 dBu signal at the TS input
Input level, max gain: 0 dBFS is obtained with a -19 dBu signal at the TS input

In terms an average human can understand, what does that mean and how useful is knowing that when setting proper levels?

Isn't 0 dBFS - digital clipping level and always stay below, and dBu is analog measurement of voltage.

What is consensus on setting up properly?

 

[PapaRaptor]

EDIT: You need not read further unless you want to. This edit was compiled after re-reading your question.

I use an AXE I/O One with the following specifications for Instrument:

Input level, min gain: 0 dBFS is obtained with a +10.5 dBu signal at the TS input
Input level, max gain: 0 dBFS is obtained with a -19 dBu signal at the TS input

It is just telling you that adjusting the gain control changes the sensitivity, as if you didn't already know that.
The actual numbers mean that the AXE I/O will produce a peak output of 0dbfs (.775 volts) over an input ranging from (-19dbu) 0.0869 volts and (+10.5dbu) 2.45 volts. If your input voltage is below the minimum -19dBu it will likely be too low to be satisfactory. If the input voltage is above +10.5 dBu, it will likely result in overdriving your input and could possibly cause damage to the input circuitry.
===========End of Edit===========


The "no math" explanation is relatively simple. All of the db references boil down to the same thing and that is keeping each processing stage out of overload (distortion).

The only places that clipping usually takes place is at the input stage when the analog input signal is converted to digital or the digital signal is rendered to the final output. Once converted to digital signalling, most (if not all) DAWs do not introduce any internal clipping.

1. If your hardware is equipped with input trimmers and metering LED's, keep them out of the red. This is really the only tricky part about it. You will need to hit the inputs as hard as you can. If the interface has peak holding indicators it makes this much easier. Hitting the red indicates you are exceeding the capabilities of the hardware digital/analog converters to do their job. Their ability to quantize a signal only goes as high as the bit count they can encode. For most hardware a/d converter interfaces this is 24 bits. If it's broke here, it's broke through the entire production process.

2. Once converted to a 24 bit fixed point digital signal (at whatever sampling rate you want (44.1, 48, 96, 192 kHz) it goes into the DAW, where it is processed at a much higher bit rate than the source material. Most modern DAWs operate using 32 bit or 64 bit floating point processing. Sorry, this does require a bit of math to explain the differences. If you're not familiar with the differences between fixed point and floating point, this does require some math explanation. I won't explain the math here, but a good analogy is fixed point is like a garden hose. It only has so much room for water (signal) to flow. Floating point changes the size of the hose as the need arises. If a signal exceeds the room allowed, floating point simply increases the size of the signal processing by an order of magnitude. There are downsides to this, but that's not within the scope of this explanation.

3. Once the signal has been massaged by the DAW and associated plug-ins, it has to be rendered into a file that is acceptable for playback. This is called rendering. This converts the processing depth (32 or 64 bit floating point) to a fixed point signal. We have to take the processed signal and squeeze it back down into something that fits in the fixed point pipeline. If the numbers coming from the floating point exceed the number of bits we can stuff into fixed point file, that's digital clipping. If you clip in rendering, it's distorted and broke. However, the good news is you can always go back and adjust the processed signal until it is no longer exceeding the size of the sample.

Higher bit depth (and sampling rate) produces higher quality audio, but it still has to pass through a d/a converter in order for you to hear it. Compact discs (CD) are standardized at 44.1kHz sampling with a 16 bit depth. Most interfaces are equipped with d/a converters with bit depths of 24 bits. The sampling rate on many is variable. The MP4 audio/video standard specifies 44.1kHz or 48kHz sampling with a 16 bit depth. However, the format doesn't preclude higher (or lower) bit rates or bit depths.

tl;dr For recording, keep as much headroom as possible. Most plug-ins work with a signal within a DAW that averages between -18db and -12db on the individual input levels. This will keep the floating point processing at a level that allows a mixed signal to have resolution and dynamic range as high as any individual input.

even more tl;dr. Keep everything out of the red.

 

[CaptainMoto]

My mathematical capacity is much more limited than @PapaRaptor's
Here's my take:
If setting the input level on your interface below clipping produces a distorted sound, you may be sending too much gain into the DAW.

To correct that try these steps.
1. On the interface > Set the input gain to keep peaks in the yellow, never red.

2. In your DAW, check the input level, set it around -18 db. This is not the track meter/slider in the DAW mixer, it's a separate view to see the DAW input levels. This can be adjusted in two ways. A) At the output of your interface or B) Within the DAW at the input adjustment, before it hits the track. Target -18dB

3. Adjust the levels on the plugin to achieve your desired tone. This will likely change the level going to your track.

4. Adjust the track slider to balance the guitar track in the mix, always below clipping. Once again, -18 dB range is a good starting point.

If you use Studio One I can point you where to check the input levels.
Otherwise google up how to adjust the input level for your DAW.

 

[TexBill]

I like the NO MATH analogy both of you wizards presented.

For me, I do not like Calculus and don't guess I ever will. Got this far down the road without having to rely on it and not planning on diving feet first into Beginning Calculus any time soon. It rests on the shelf behind me if a reference is required.

So, from the standpoint of gain, dB is a good yard stick. I agree with the -18 dB to - 12 dB levels suggested. The amount of input gain is the driving force and is easily controlled....

Kudos to both PapaRpator and Moto for the explanation on how each element affects the outcome.....

 

[CaptOblivious]

Much appreciated @PapaRaptor and @CaptainMoto for your response!

Interestingly, checking my input level in Reaper found setting Axe I/O One all the way to the left - minimum - was peaking at - 16dB, when I set the gain to be just below clipping on the Axe in Reaper was at -5dB. On the AXE itself with the input gain all the way to the left, the meter was maxing out at -12dB

Playing in a digital world only, I went back to YT to see the videos about setting input level, many were too technical for me, hence my original post. I came across this guy who posted this two weeks ago, he mentions pretty much everything I was having problems with. Unheard rule of Dialing in your amp sims and this one is the other video he mentions, Ed S how to set optimal input level.

For Amplitude users, Are you using Amplitude wrong

I will say, though, once again spending way too much time trying to dial in tone and not enough counting.

Suspect I am conclusively proving tone (mediocre in my case) is in the fingers

 

[CaptainMoto]

To me, complicating things with dBu is like trying to drive a car under the speed limit while looking at the tachometer rather than speedometer.

I’ll have to watch those YTs to see what the heck they are talking about

 

[CaptainMoto]

OK,
I watched the YTs, I see what they are saying.
Setting aside the recommended specific level for each plugin, it's all basic gain staging.

Along the signal path there are several points where you can adjust the gain.

For a guitar into an amp, you can adjust the gain at the volume pot on the guitar and at the preamp on the amp.
If you add any effects pedals each one is an additional gain stage.
Consequently each stage is sent to the next and the gain is stacked giving a final resulting tone.
Guitar Volume > Pedal Gain > Preamp

When using plugins or amp sims, it's the same thing with a few extra stages.
The first stage is the guitar volume, followed by the interface gain, that is sent to an input within the DAW.
Each input also has an output gain so, before it even gets to the channel you have two points (in the DAW) at which gain can be adjusted.
The channel has an input level and an output level.
The channel input is the amount of gain hitting the plugins.
The output (represented by the channel meter) is the net result of all the input stages, as well as, any gain added by plugins.


So, there are several stages at which gain can be changed.
Guitar Volume > Interface gain > DAW Input level > > Channel Input level > Plugin input gain > Plugin output gain > Channel Output gain > *Output level

If you want to achieve a recommended input gain feeding the plugin there are four points along the path that can be adjusted to accomplish the objective


I'm thinking, if you set the input gain low on the interface that's one approach but.................Maybe not the best
If your interface introduces any noise ( high noise floor) that noise will be amplified when you add gain anywhere in the DAW.
So, back to my earlier recommendation, set the interface gain just below clipping and if that's too hot for your amp sim, adjust the gain down at one of the points before it hits the channel.


To check the noise level coming from the interface just look at the input signal without playing the instrument.
Then as you dial up the plugin to the desired level see how much the additional noise is represented on the channel meter.

*Output level is typically set as your MAIN or Master Bus which will be the value of the entire mix but, can also be changed to other outputs such as outboard gear ( like EQ or Compressor) which come before it hits the Mian Output

 

[PapaRaptor]

I watched (as of right now) up to about 7 minutes of the Sebastian Sampson video and I have to disagree with his advice about setting the initial gain on the interface input.
The interface probably has multiple connection capability for an input. Usually when they do (such as a combo XLR and TRS), the XLR is expecting the lowest signal (mic level), while the TRS is expecting a line level. There may be a switch to select an instrument level (which is higher than Mic and lower than line). Further, there is no guarantee that an XLR connection is going to be at mic level. It's a rule of thumb, but no guarantee.

So, unless you know exactly which connector is expecting which level signal on your interface and what the level may be coming from your source, setting your preamp to minimum gain may work, but probably isn't the best choice.

The analogy referred to in his "why don't I just dime the input of the interface?" is flawed. It is NOT like putting a tube screamer in front of your amp. Most "real" guitar amplifiers are analog circuits. The input gain on an interface is setting the level of signal that will be fed to the a/d converter. While both the tube screamer and overloading the d/a converter will result in distortion, the two are not even remotely comparable.

No matter what bells and whistles your interface has on the front panel, it's main purpose is to do one thing and that is convert your analog source into a digital signal, usually 24 bits deep. The input source adjustment sets the input analog stage to properly supply the a/d converter with a signal.
If the signal is too low, the converter isn't going to utilize enough of the bit depth to articulate proper resolution. In human speak, this means you're going to have a low digital signal to process. If you need to digitally turn it up your signal is going to become more grainy (noise).
If the signal is too high, the a/d converter will simply go to the top of its bit depth for everything that is too large for it to capture. This makes for really ugly distortion and is to be avoided at all costs.
So, in my opinion, you want to make sure that your input signal falls into the baby bear range, where it's not too hot and not too cold. You want to make full usage of the 24 bit depth of the converter by feeding it enough signal to exercise the maximum range within the converter. This ensures that you can achieve the full dynamic range of your instrument in the recording.

Once your signal is cleanly into the digital realm, then by all means, the instructions given in the video are as valid as any of the other recording "engineers" who espouse on Youtube.

 

[PapaRaptor]

On the same subject, here's an explanation that is at odds with what I previously said. I present this because I am not an expert on the subject.

I Am A Little Confused About This…

Recently I’ve seen some confusing info on using guitar plugins. So I wanted to help you guys by showing you some useful info that should clear things up a bi...

youtu.be

If you're not familiar with the person in the video, he has been a regular in some of the Anderton video reviews and is a metal guitarist of note in the U.K and Europe (possibly in the U.S. as well).
.
He pretty graphically demonstrates the idea of keep your interface lower if you're using amp sim plug-ins inside the computer. This makes a pretty good case for not necessarily following my suggestions in previous posts if you're using amp sims.

 

[CaptainMoto]

I'm holding fast to my previous comments:
The vid talks about the input needing to to match the requirements of the amp sim.......Great, I agree!

He goes on to suggest that the way to do it is by adjusting the input gain on the interface.
I say, that's one way and there are several gain stages along the way that need to be taken into consideration.

I still believe, it's best to set the interface input gain at a level below clipping and make the necessary input gain to suit your amp sim inside the DAW rather than dialing down the interface. That could be at the DAW input or possibly at the channel input level.

It's the amount of gain hitting the sim that matters and how you get there is irrelevant. My reco would be set the interface at a good level and adjust the gain inside the DAW.

I can't be wrong because I am an expert!

 

[CaptOblivious]

On the same subject, here's an explanation that is at odds with what I previously said. I present this because I am not an expert on the subject.

I Am A Little Confused About This…

Recently I’ve seen some confusing info on using guitar plugins. So I wanted to help you guys by showing you some useful info that should clear things up a bi...

youtu.be

If you're not familiar with the person in the video, he has been a regular in some of the Anderton video reviews and is a metal guitarist of note in the U.K and Europe (possibly in the U.S. as well).
.
He pretty graphically demonstrates the idea of keep your interface lower if you're using amp sim plug-ins inside the computer. This makes a pretty good case for not necessarily following my suggestions in previous posts if you're using amp sims.

I have seen this, and there are other YT'bers with big followings saying the same thing. Rhett Shull demonstrates A/B's actual amp and plugin before and after A/I adjustments. It's a pretty long video, fast-forward a bit to get to the comparisons, pretty conclusive. The Neural DSP folks have changed their plugin manual to suggest leaving A/I gain at minimum and adjust the plugin slider, how much you adjust depends on your device.

The THR folks say "However, preamps affect the quality of the audio and can also add character to it, and in some cases, you might want/need one depending on what kind of sound you’re going for", which is the issue for plugin users who want their plugins to sound like the real thing. The problem for manufacturers is there are a lot of interfaces, all with different specifications. It may also explain why those youtubers say their tones sounded like a tube screamer was built in.

In my case, living in a digital world, setting the A/I gain the traditional way was too hot for Amplitude. I could never get a decent clean tone, drive and hi-gain tones pretty much sounded wrong, thin and screechy. Hopefully @PapaRaptor ears have stopped ringing after one of my early VJR postings blasted them. I could not tell you what a 59 Bassman or 65 Super Reverb sound like in the real world, only that what I was hearing sitting at my desk in my office with headphones on the "tone" sucked setting up the "traditional" way. Since seeing those videos I have been following their advice and can say, to my ear, my tone sucks a lot less. It really is the fingers. The IK Multimedia folks don't make it easy, seems every amp collection they sell are made differently and do not tell you how, leaving their users to guess what the proper matching gain should be.

Beyond "stay out of the red" in the digital only world there are multiple ways to set your tone, it really all comes down to what sounds good to you.

 

[PapaRaptor]

I agree completely with @CaptainMoto's post.

Pretty much since I got the Eleven Racks back in the stone age, they have been my input source. When I'm recording a guitar or bass, it is usually through an 11r. Certainly, I have been spoiled, since the Eleven Rack provides a processed signal (mine is usually set to Super Reverb) with a little bit of stink added by a TS emulation and a touch of plate reverb. It also simultaneously provides a totally dry signal. While I can control the input levels of the processed guitar (or bass), the dry signal offers nothing in the way of a gain trimmer.

The way my audio system is set up, I hear the processed guitar signal during recording, independent of the input levels on the DAW, so monitoring through the DAW has never been an issue. In spite of that, the processed track is often discarded (or at least muted) and I use the dry track. Studio One has a macro included in the base package to normalize a recorded signal to -12db, which I usually use on the dry track, prior to passing it through Ampire (Presonus' modeling module) or Bias FX (which I really don't use that often). The normalization macro doesn't significantly impact the dynamic range of the guitar, but it gives me a level playing field for everything I record.

The idea of exactly matching an amp sound has never been on my list of things to do. I just want it to be loud and most of the time, be relatively clean. I don't think I've ever tried to nail down a specific amp. If I have, it was a long time ago. That certainly affects how I approach the recording process. If you are "amp chasing," then your course of action may be different and the suggestions in the Youtube videos are certainly something to consider.

 

[CaptainMoto]

I watched several vids on the subject and they pretty much all say you need to reduce the input gain to match the required level in the amp sim.
NO NEW NEWS!

What is new news to me is............
I have never used Reaper so I watched some tutorials and from what I see, Reaper does not have a seperate input level display or adjustment, It all occurs at the track level.

I use Studio One which does have that feature so, my example of adjusting input gain "in the DAW at the input" is not available in Reaper.

So, looks like the only way to effect input gain in Reaper is at the interface.
My apologies for adding confusion to this discussion.

I tested my theory in Studio One with an amp sim and found it works as advertised when I leave the interface gain set at a normal level and dial it down at the DAW input stage, before it hit the channel.

I shall go silent on this now............I'm sure that will make some of you happy

 

[PapaRaptor]

I shall go silent on this now............I'm sure that will make some of you happy

Nah, not happy. But I think we have beaten this to death.

 

[CaptainMoto]

Nah, not happy. But I think we have beaten this to death.

I'm always learning.
This has been a good learning opportunity

 

[PapaRaptor]

I'm always learning.
This has been a good learning opportunity

Yes, I learned the differences between 0dB 0dBu, 0dBv and 0dBfs and also realized why calculus and I didn't gel.