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Unity Gain for amp blocks.


CraigGT
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I first saw this mentioned by Ben Adrian and have seen a few mentions of using the amp channel gain to make sure that the amp block is not adding any significant gain to the path. Then when I was building the Lonestar manual presets I could clearly hear distortion on the clean sounds and found this to be the cause, reducing the channel volume cleaned them up.

 

That got me thinking that by default most amp blocks are set much higher than unity and is this the cause of the digital distortion that people complain about and even the problems I had overloading my Strymons in the loops?

 

Clearly in my case something was distorting downstream from the amp block and it seems that it would be a trivial matter to set channel volumes more appropriately.

 

Craig

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I'm also one of the "weird people" that go for unity gain on every block such that the total patch loudness is the same as a blank patch. Except for optional solo boosts of course. Strange considering a benefit of digital being consistent processing at any level (within the device's operating range).

 

Unfortunately, for dynamic blocks like amps, dirt boxes and compressors, there is no way to set a single default level for every guitar. Pickups can have very different outputs.

 

E.g. An amp is set to unity gain for a vintage output guitar.

Bypass lvl = active lvl = reference 0dB.

 

Switching to a hotter output guitar (say +12dB), bypass lvl = +12dB.

But with the amp active, there is more clipping than before. Output lvl compressed to say +6dB.

Now bypass lvl > active lvl. No longer at unity gain.

 

Short story, you can only adjust to unity gain for 1 guitar at 1 pickup and volume knob position.

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I found a problem literally the other day where one of my cleaner amps had some crackly top end that I had initially put down to me not understanding how the amp was meant to sound - but it went away entirely when I turned the amp channel and/or master volume down. But there was no other indication that I was overloading any of the blocks and the final output level was way below 0 dbFS.

 

Line 6, can we please, please get some sort of clip/high signal indicator on each individual block?

 

As for the unity gain idea, I don't think it's very practical. An amp block should amplify the signal - the hint's in the name. Even if it can be persuaded not to do so, a cab block following it is almost always going to significantly attenuate the signal. If I have the 2 blocks next to each other then I try to make sure that what comes out of the cab is not much different from what goes into the amp, but I think there needs to be better metering for all of this.

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As for the unity gain idea, I don't think it's very practical. An amp block should amplify the signal - the hint's in the name. Even if it can be persuaded not to do so, a cab block following it is almost always going to significantly attenuate the signal. If I have the 2 blocks next to each other then I try to make sure that what comes out of the cab is not much different from what goes into the amp, but I think there needs to be better metering for all of this.

 

If this was the real world then the signal coming out of the mic would actually be lower than that of the guitar going into the amp but that's just being pedantic.

 

It does seem like a real issue though that could be avoided with a bit of metering or maybe just making a block flash if it is being hit too hard in the digital domain.

 

Craig

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If this was the real world then the signal coming out of the mic would actually be lower than that of the guitar going into the amp but that's just being pedantic.

 

It does seem like a real issue though that could be avoided with a bit of metering or maybe just making a block flash if it is being hit too hard in the digital domain.

 

Craig

 

I'm not entirely sure what you mean by that first sentence, but I think it reinforces one of the issues here.

 

Basically, we have a system which allows us to arbitrarily rearrange blocks in whatever order we like, but those blocks sometimes simulate real-world hardware that is very sensitive to the absolute input level - for example, amps which only start to break up or saturate when the signal is hot enough. This alone tells us that we shouldn't expect there to be one 'true' signal level that is going to work across every block throughout the whole chain, and that in turn means that striving for unity gain on each block is only going to work if we know that every block is calibrated broadly the same and that we're getting enough signal in the first place.

 

Some of this would be helped by there being better documentation from Line 6 regarding expected levels... with my programmer hat on I assume it's all floating point data in the pipeline so you don't get any hard clipping before it hits the output, but it certainly does have some nasty sounding distortion in some blocks when the level is too hot. So if that's part of the model, let us know so we can attenuate accordingly.

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I'm not entirely sure what you mean by that first sentence, but I think it reinforces one of the issues here.

 

What I meant was that you put an instrument level signal into your amp, the amp puts a massive speaker level signal into your cab, then your good old SM57 puts a mic level signal into the desk. so the output of the complete system is actually lower than the input.

 

Craig

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Well, maybe! Depends on the mic, depends on the gain on the desk, etc. But yeah, this is my point - we're simulating hardware and parts of the chain that intrinsically amplify and attenuate the signal. So the concept of 'unity' gain implies counteracting some of those changes which in turn is going to affect certain sonic characteristics of the blocks that come later in the chain.

 

I don't think there is any "one size fits all" solution here but the fix would be (a) having guidance from Line 6 on what the input level to each block should be, and (b) getting realtime feedback from the unit on whether we're successfully doing that or not.

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Line 6, can we please, please get some sort of clip/high signal indicator on each individual block?

 

Hear, hear! Would like more visual feedback in general. Would also like to be able to see when the threshold on the compressors is engaging and for how long as well as see the parametric EQs display a graphical curve which the Global EQ already has. A Real Time Analyzer(RTA) would be icing on the cake.

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Hear, hear! Would like more visual feedback in general. Would also like to be able to see when the threshold on the compressors is engaging and for how long as well as see the parametric EQs display a graphical curve which the Global EQ already has. A Real Time Analyzer(RTA) would be icing on the cake.

All of that would be fantastic "honestly."

(pun intended) But I don't see all of that being implemented, though I would love to be wrong about that. 😊

 

Though I think that clip indicators on each block would be a tremendous move in that direction, maybe a bright red box flashing on the block that has a clipped input, and a yellow box flashing on the block that has a clipping output. I would be satisfied with that, but the other stuff you mentioned would be cool too.

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All of that would be fantastic "honestly."

(pun intended) But I don't see all of that being implemented, though I would love to be wrong about that.

 

Though I think that clip indicators on each block would be a tremendous move in that direction, maybe a bright red box flashing on the block that has a clipped input, and a yellow box flashing on the block that has a clipping output. I would be satisfied with that, but the other stuff you mentioned would be cool too.

 

I think you're right with a clipping indicator being lower hanging fruit and more likely to be implemented. I don't see any reason though why at least a simple indicator of when the compressor is kicking in and in operation couldn't be added as well as adding the same graphical curve display to the parametric EQs that already exists on the Global EQ(which after all is just a PEQ applied globally). The RTA is probably the most pie in the sky item on the wish-list as it might be the most difficult to implement and I don't know how much DSP it would require. Sure would be handy though.

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Btw, as others have mentioned, even if some of this functionality can't be placed directly on the Helix there is no reason it could not be placed in the Editor. There is no reason the editor has to have exactly the same functionality as the HX devices it serves even if there is a tight linkage. I am not proposing a standalone editor which would introduce issues regarding synching and latency between for example the Helix and the editor. There are also the ever-present DSP limitations when you add something on the device. Just talking about using the editor to add some capabilities that at least initially might be more difficult to place directly on the device and might be much simpler to deliver given a computer's potential resources. This additional functionality would be mighty welcome even if it was only available via HX Edit. Especially since as has been noted elsewhere, most of the tweaks done to a preset are executed before hitting a stage and often done when an editor is available.

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I think you're right with a clipping indicator being lower hanging fruit and more likely to be implemented. I don't see any reason though why at least a simple indicator of when the compressor is kicking in and in operation couldn't be added as well as adding the same graphical curve display to the parametric EQs that already exists on the Global EQ(which after all is just a PEQ applied globally). The RTA is probably the most pie in the sky item on the wish-list as it might be the most difficult to implement and I don't know how much DSP it would require. Sure would be handy though.

Other than the clip indicators I could really use the threshold indicator you speak of for when the compressor kick in.

Those are the two that would help me most. The EQ display would be very helpful too. The RTA would be least helpful for me personally, as I do all my work in my DAW, and have plenty there. So I am kinda glad it is the least likely to be added of the above suggestions, but I wouldn't be upset if they did add one. 😉

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Using Helix with DAW metering through the USB interface can be quite helpful. Or you can use the input and output meters in Helix Native to gain stage your patches. By selectively bypassing blocks, you can get a pretty good idea how the patch as a whole is gain staged.

 

This is important because each block is designed to work best with a specific input range. In the absence of any other information, unity gain is a pretty safe starting point - each block sees essentially the same input signal level.

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Other than the clip indicators I could really use the threshold indicator you speak of for when the compressor kick in.

Those are the two that would help me most. The EQ display would be very helpful too. The RTA would be least helpful for me personally, as I do all my work in my DAW, and have plenty there. So I am kinda glad it is the least likely to be added of the above suggestions, but I wouldn't be upset if they did add one.

 

I hear you. RTA standalone apps and plugins are available for your DAW and do provide the core functionality needed.  It makes sense to me for Line6 to de-prioritize feature requests that are available via other avenues or would require too much development time or DSP. This is to some extent an ease of use issue, no doubt there are other alternatives available. For me there is just no substitute though for an RTA that pops up right behind the controls of a PEQ or a GEQ. Having for example an RTA that displays behind the 31-band graphic EQ(s) is a feature I look for in any mixer. Even though we are not dealing with 31-frequency bands I can see how useful it could be with a Helix.

 

When the RTA appears directly behind the graphical curve of your parametric or graphic EQ it becomes child's play to precisely nail the frequencies you are trying to adjust.  I find this much easier than trying to flash back and forth visually between, for example, the editor and the marked measurements on a DAW's separate RTA plugin. Having one right on the Helix screen or Editor would be very informative and helpful for shaping tones and finding problem spots. That and my ears. Not expecting this any time soon. Just one of those things that I think is coming to guitar modelers eventually, if not in this version of the Helix than maybe a later one.

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When the RTA appears directly behind the graphical curve of your parametric or graphic EQ it becomes child's play to precisely nail the frequencies you are trying to adjust.

Oh of course. One of the many reasons I love my Fabfilter Pro Q2. Visual aids are very helpful even if the ears are the final judge.

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What if there was a device that could generate a reference noise that is sent to the Helix, and then a return block inserted as desired in the chain returns to the device which shows the amount of gain relative to the reference noise?  Or do I misunderstand how signal metering works?  Is there equipment around that can do that?

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What if there was a device that could generate a reference noise that is sent to the Helix, and then a return block inserted as desired in the chain returns to the device which shows the amount of gain relative to the reference noise?  Or do I misunderstand how signal metering works?  Is there equipment around that can do that?

 

Any basic mixing board will do that whether it's in a DAW or a hardware mixer.  I personally use a small Alesys 6 channel mixer and use a DI box to route my 1/4" out to both my speaker and the mixer (the mixer isn't connected to any speaker).  That allows me to use the signal meters on the mixer to help me normalize all of my signals to a consistent level to make it easier for the sound man to gain stage me at sound check time.  This ensures all my patches and snapshots stay consistent throughout the performance to reduce any risk of clipping.

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Any basic mixing board will do that whether it's in a DAW or a hardware mixer.  I personally use a small Alesys 6 channel mixer and use a DI box to route my 1/4" out to both my speaker and the mixer (the mixer isn't connected to any speaker).  That allows me to use the signal meters on the mixer to help me normalize all of my signals to a consistent level to make it easier for the sound man to gain stage me at sound check time.  This ensures all my patches and snapshots stay consistent throughout the performance to reduce any risk of clipping.

 

I have an audio interface so, with everything at 12 o'clock on the interface (I'm assuming that would be unity), I could put the default 3 note generator block (for consistency) in front of each preset that I want to balance and then use software meters to dial in the same level across the presets I want to balance.  Would that work?

 

Would it be worth putting a return block after the amp block and use the channel volume to balance in addition to balancing the final output?

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The Helix would benefit from the addition of some metering.

 

 

Line 6, can we please, please get some sort of clip/high signal indicator on each individual block?

 

 

This! 

 

A good example is the Eventide H9 (a great pedal, BTW), which has meters expressly in order to 1) Not overload the input, and 2) Send unity gain back to the amp or fx loop.  It works well, and it's super easy to achieve unity gain through the H9 effects.  In fact it has optional automatic unity gain matching, so if you raise the input level, you can have the pedal set the output gain if you wish.

 

Helix is very advanced, but the comments about different guitar pickups is true ... they push the inputs in a huge variety of unpredictable ways, and input meters would make it dead simple to avoid clipping, as well as optimize input signal level.  I would love (at the very least) input and output meters.

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I have an audio interface so, with everything at 12 o'clock on the interface (I'm assuming that would be unity), I could put the default 3 note generator block (for consistency) in front of each preset that I want to balance and then use software meters to dial in the same level across the presets I want to balance.  Would that work?

 

Would it be worth putting a return block after the amp block and use the channel volume to balance in addition to balancing the final output?

 

Speaking from the perspective of a sound man, unity is only important in terms of the setting of the faders for each channel.  If gain staged appropriately the signal coming into the board will be mixed appropriately with the other channels and have ample headroom to account for momentary peaks as well as possible momentary adjustments on the faders if necessary to enhance the mix.  That is literally the only thing important in live sound as it relates to unity.

 

If you have your master volume knob set to noon, and you're making normal adjustments to channel volumes and not going crazy with additional volume and gain adjustments, you have more than enough headroom to ensure your signal won't clip coming out of the Helix.  The higher you turn up your master volume the less headroom you will have and the more potential you will have for clipping the signal at the Helix.  What IS important is that each one of your patches and snapshots present a consistent signal level to the mixing board so they can gain stage that signal one time at sound check and be assured of some level of consistency. And the only way to do that is to play normally through each patch or snapshot at your preferred normal master volume setting while connected to some from of signal meter and adjust your channel volume or channel output to present a fairly constant level so it can be gain staged consistently at the mixing board.

 

In my case I have my Helix master volume set at 11 o'clock.  I have the gain knob on my mixing board channel set to unity or 12 o'clock.  I adjust all of my patches so the signal level measured at the board is -6db normally with intermittent peaks going briefly to -3db.  Given that most boards clip at around +6db, that ensures that I'll have that same amount of headroom on my 1/4" output going LINE level to my FRFR speaker (which is set at 12 o'clock or unity), and on my XLR output going to the board at MIC level at maximum output since I have the master volume disabled on the XLR output.  That leaves more than enough headroom to ensure the Helix won't clip any signals, and the rest is up to the sound man to dial in the trim or gain knob on the mixer to put my signal where he wants it to be.  That's ALL that really matters....NOT unity.  Unity is something the sound man will deal with.

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In my case I have my Helix master volume set at 11 o'clock.  I have the gain knob on my mixing board channel set to unity or 12 o'clock.  I adjust all of my patches so the signal level measured at the board is -6db normally with intermittent peaks going briefly to -3db.  Given that most boards clip at around +6db, that ensures that I'll have that same amount of headroom on my 1/4" output going LINE level to my FRFR speaker (which is set at 12 o'clock or unity), and on my XLR output going to the board at MIC level at maximum output since I have the master volume disabled on the XLR output.  That leaves more than enough headroom to ensure the Helix won't clip any signals, and the rest is up to the sound man to dial in the trim or gain knob on the mixer to put my signal where he wants it to be.  That's ALL that really matters....NOT unity.  Unity is something the sound man will deal with.

 

My setup is a little different. I have the Helix output volume knob set to control the 1/4" outputs for my backline FRFR. The XLR outputs are fixed and maxed so I send a consistent, independent signal to the FOH mixer. We use a digital mixer, so I set the input gains closer to -18 to -12 dB. This ensures no digital clipping in the channel strip. Then i start with the channel faders around -18 dB (mid scale) so that there's plenty of room to adjust the mix, and the sum of the channels doesn't clip any of the output or master buses. Headroom is pretty important in a mixer, and an absolute must for digital mixers.

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My setup is a little different. I have the Helix output volume knob set to control the 1/4" outputs for my backline FRFR. The XLR outputs are fixed and maxed so I send a consistent, independent signal to the FOH mixer. We use a digital mixer, so I set the input gains closer to -18 to -12 dB. This ensures no digital clipping in the channel strip. Then i start with the channel faders around -18 dB (mid scale) so that there's plenty of room to adjust the mix, and the sum of the channels doesn't clip any of the output or master buses. Headroom is pretty important in a mixer, and an absolute must for digital mixers.

 

Actually I wasn't talking about our live mixer.  I was talking about my testing mixer at home which I use to normalize my signal levels across my patches and snapshots, so that level is somewhat arbitrary.  This just ensures I'm sending ample consistent signal strength so the sound man can gain stage it in any way he wants for his equipment.

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While I was reading this thread.

 

Does anyone know for sure whether the Channel Volume Parameter in the Amp Model does the same as the Microphone Level Parameter in the Cabinet Model?

 

Why do I turn this issue here?

 

With utmost respect to a unity level approach in Helix I'm still always trying to reflect analog realities. So, I remember that some speakers - especially Greenbacks - do perform better, when they are operated close to their limit. That's why many Guitarists used to run their Greenbacks very loud thus challenging FOH and studio guys. The only way to cope with this, was - if not sending the Guitarman home - to turn down the Microphone level, especially when you really built Signal chains in which Compression, Modulation, Delay and Reverb came after Cabinet and Microphone.

 

Now, if the Channel Volume Parameter really has NO influence on the Cabinet tone, on the Mike Model behaviour etc. it might be useful for a unity level approach, but if it has, you should better use the Microphone Level Parameter in the Cabinet Model.

 

[EDIT] I created several test presets in which I compared higher/lower Channel Volumes (Amp Model) versus lower/higher Microphone Levels (Cabinet Model) at a few different Amp, Cabinet, Microphone Models and Parameters. I didn't find audable differences, but I'm not sure.

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While I was reading this thread.

 

Does anyone know for sure whether the Channel Volume Parameter in the Amp Model does the same as the Microphone Level Parameter in the Cabinet Model?

 

Why do I turn this issue here?

 

With utmost respect to a unity level approach in Helix I'm still always trying to reflect analog realities. So, I remember that some speakers - especially Greenbacks - do perform better, when they are operated close to their limit. That's why many Guitarists used to run their Greenbacks very loud thus challenging FOH and studio guys. The only way to cope with this, was - if not sending the Guitarman home - to turn down the Microphone level, especially when you really built Signal chains in which Compression, Modulation, Delay and Reverb came after Cabinet and Microphone.

 

Now, if the Channel Volume Parameter really has NO influence on the Cabinet tone, on the Mike Model behaviour etc. it might be useful for a unity level approach, but if it has, you should better use the Microphone Level Parameter in the Cabinet Model.

 

[EDIT] I created several test presets in which I compared higher/lower Channel Volumes (Amp Model) versus lower/higher Microphone Levels (Cabinet Model) at a few different Amp, Cabinet, Microphone Models and Parameters. I didn't find audable differences, but I'm not sure.

As long as you're within headroom limits, there's no difference. Impulse response convolution is a linear process. If you want a pushed speaker sound, you have to use an IR that was captured with a pushed speaker.

 

For a lot of applications, there's absolutely no problem with mimicking analog signal chains and only balancing the final output volume.

 

The problems arise when the patches get complicated. Say there's a snapshot where the amp and IR are turned off, acoustic IR turned on, for acoustic sound with piezo. Then you gotta use snapshots to control levels. Add more complicated switching in the other snapshots and things get messy fast.

 

Then if I want to manually bypass a block in a live situation, there's a high chance for the levels to jump suddenly and take people's heads off. Good luck dealing with the soundguy after that...

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Line 6, can we please, please get some sort of clip/high signal indicator on each individual block?

 

 

What would be helpful is for someone of authority from Line-6 to say whether or not this is even possible to do with the current Block Icons in Helix. And if so, then we go from there.

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