amsdenj

So 0 to +7dB is what I'm suggesting you avoid. Rather gain stage for unity gain though out your signal chain, ensuring your loudest patch is not clipping Powercab input. Then go to the output block and set a controller to min -3dB, max 0. Set your stage volume and FOH volume with the output set to -3dB (boost off). Then when you boost to 0dB, you're back where you started from with your gain staged signal path, no chance for digital clipping, and you're 3dB louder. Same outcome, but less chance of clipping anything. Of course FOH will need to gain stage similarly. I think of it as "push tone, pull volume". That is, all you're up front patch parameters should focus on getting the tone you want to deliver to the final output stages. Then you get the venue volume by turning up the power amps at the end of the signal chain instead of turning up volumes in the front or middle of the chain as these consume headroom. Want the PA louder, you turn up the master first, not all the faders. Turning up the faders can clip the master bus, turning up the master fader can't.

Reactive load boxes with IR capability and direct FOH outputs are breathing new life into typical 100W tube guitar amps. If you love your amp, and can afford to lug it around, then this is a great solution. But I think it may have a limited life. I'd much rather use Helix and Powercab, more flexibility for less weight and footprint.

Unity gain is the output is at the same level as the input. That will be the case for a Helix patch that has no blocks, assuming the output block level is set at 0dB. Now as you add blocks to the patch, they may add or remove gain depending on the block. To gain stage a block to unity gain, adjust the block to get the tone you want, then use the block's Level control to make the overall volume the same with the block on or off. This doesn't need to be exact, just close. If you do this, you will 1) provide consistent inputs into downstream blocks, 2) avoid any possibility of digital clipping, 3) drive blocks with the levels they were designed for, and 4) have consistent output to your Powercab, FRFR or direct FOH from patch to patch.

Snapshots and stomp switches give Helix a lot of flexibility. Snapshots can store values for up to 64 parameters in a patch. stomp switches can control up to 8 parameter min/max values or bypass state. The big difference between the two is that snapshots aren't additive - you can't turn on snapshot 2 and 3. Stomp switches are additive, you can have any combination of up to 10 stomp switches controlling up to 8 parameters/switch on or off at a time - that's a lot of tonal combinations. So should you use snapshots or stomp switches? It depends on the use case. If you're playing guitar in an orchestra pit for a dance performance or play, then you're probably playing a fixed sequence of songs with fixed parts in each song. The critical thing is to play the right sound at the right time, following the conductor. Having a preset per song, and snapshot per song part is very convenient for this situation, and can significantly reduce the chance of getting the wrong sound. Contrast with a jamband in a club where the song order is unpredictable, and solos can start anytime and go on for a while. In cases where there's less structure in the performance, having a palette of stomp switches for overdrive and other effects lets you control the tone for the moment, turning on and off combinations of effects. I play in a rock cover band. We plan a set list, but don't always follow it depending on how much people are dancing - we really try to keep people in the club dancing. So there's some structure in the song sequence, and we do have arrangements for each song. But I prefer to use stomp mode where I have my typical sound and effects in a patch I use most of the time. I like this because I don't always play the song exactly the same way - I might use a Mutron instead of a Wah for example. I have a few patches for odd things that are unique to the song. For example, we do Love The One Your With and I use a 12string opened tuned acoustic (Variax) through an acoustic patch with a snapshot for the EBEEBE tuning. I also keep an open-G tuning snapshot in my regular preset for Stones songs.

You are probably getting a mix of direct and software monitoring. Use direct monitoring on your interface if it supports it, and you don't need any effects on vocals while recording. Use software monitoring otherwise (turn any direct output to "computer"). I use software monitoring exclusively because of the flexibility, and the ability to run amp models like Helix Native while recording a dry guitar signal. But you need a good computer and audio interface to keep the latency low.

Also make sure you're using a High-Z input in your audio interface. Electric guitars don't respond well to Lo-Z.

Excellent advice. Reaper is also cross-platform, Mac and Windows, making it easier to collaborate with others remotely through Dropbox or (now) iCloud. This is something we're probably going to be doing for a while with all the stay at home restrictions. Reaper is also very extensible and has UI skins for different looks. I'm on a Mac and use Logic Pro X. Its UX, work flow, Drummer and software instruments make it a good choice for Mac users. But I have used Reaper a lot and still collaborate with others using Reaper.

What I like about Derailed Ingrid is that you can really distinguish preamp vs. power amp clipping. Keep the Drive below .5 and any clipping is from the power amp controlled by the Master volume. This is nice, crisp, aggressive power amp distortion with odd order harmonics that we’re all pretty use to. But if you bring the Master down and the Drive up, the clipping comes from the preamp which is clipped at the top and rounded at the bottom, just like a triode tube’s cutoff and saturation. This is asymmetric clipping and has more even order harmonics. I like Litigator too, but it doesn’t behave as much like a traditional tube amp as Derailed Ingrid. It has some distortion even at really low Drive and Master levels and the clipping doesn’t look exactly like what would be produced by tubes. Maybe that’s because of the gain staging and position of the tone controls in the circuit. Or maybe its just some Ben Adrien magic. I tend to switch back and forth between these amps never deciding which one I like best. They’re both wonderful.

Any block that has a mix control doesn’t necessarily need to be in a parallel path since that’s effectively that the mix control does with the wet and dry signal. However there may be other reasons for putting things on a parallel path depending on what you want feeding into what. The classic example is delay and reverb. Do you want 1) to delay your reverbs (delay block after reverb) or 2) reverb your delays (reverb block after delay), or 3) neither (reverb and delay on parallel paths and mixed later). There’s another consideration for flangers - do you want flangers before or after distortion (distortion blocks and amp block). These will sound quite different because flanger before distortion will get clipped and impacted by the distortion voicing. Flanger after distortion won’t get clipped and the high end created by the regeneration control won’t be cutoff. These are both valid scenarios and work for different purposes. You’ll have to try each to see what works for you and the song. Note however that we are use to hearing these effects in front of a distorted amp, so that might be a good place to start.

Very interesting. Thanks for pointing this out. That's a pretty sizable scoop. Did you try Derailed Ingrid? My explorations shows that model behaves more like a real amp then some of the other models.

Reaper would be a good choice. You don't have to use it as a DAW, you can just treat it like a digital mixer that supports plugins. And its not that expensive.

I have indeed read https://robrobinette.com/Tube_Guitar_Amp_Overdrive.htm many times.

Do you mean amp frequency response not impedance curve? A speaker's impedance curve is how its impedance changes based on frequency and load - basically because of stretching the cone and spider and moving the coil into and out of the magnetic field of the speaker magnet. A speaker's impedance is specified as a nominal value, usually 8 or 16 ohms, but can vary widely under load. The power amp's damping factor and negative feedback attempt to account for changes in speaker impedance to provide a more linear response. When amps distort, negative feedback is lost and the damping factor decreases. Change in speaker impedance under load is not generally captured directly an impulse response. S-Gear convolver has a Z control that attempts to model this. Redwirez BigBox IRs also come with impedance curve IRs that can be mixed with the speaker IRs to model this effect.

It occurs to me that a lot of readers of this forum might not know what bias actually is and why they should care. Changing the Bias and Bias X controls on a Helix amp block might not do that much to the tone, depending on how hard the amp is pushed. Understanding what bias is might help you understand what these controls do. Here's a brief description of bias that might provide a context for understanding what it does in Helix. Tubes amplify by heating up a cathode that boils off free electrons. These electrons pass through a grid and are attracted to a positive plate, creating a current in the tube. A small change in voltage on the grid can effect a much larger change in voltage on the plate which is an amplifier. Now imagine an alternating input voltage on the grid. On the negative half of the input voltage, electrons will be repelled by the grid back to the cathode, reducing the current through the tube. On the positive half of the input voltage, electrons will be accelerated past the grid to the plate, increasing the current through the tube. This is how a tube amplifies an input signal. Now a tube has operating limits based on how it is designed - how hot the cathode can get, how big the plate is, how far away from the cathode it is, etc. If you keep increasing the negative voltage on the grid, eventually the electron flow will stop, and at that point increasing the input further results in no corresponding further decrease in the output. This is called cutoff. Its very sharp, and results in a lot of odd order harmonics. If you increase the positive voltage on the grid, it will attempt to push electrons faster and faster to the plate. But eventually the plate won't be able to consume any more of the electrons and the output won't continue to increase with the input. This is called saturation. Saturation distortion is more gradual, less sharp and generates even order harmonics. Bias is a voltage difference between the cathode and grid that establishes the operating point of the tube between saturation and cutoff. Normal bias sets the voltage so that the tube is operating over its linear region. Hot bias reduces the bias voltage, allowing more current to pass through the tube when there is no input - the quiescent current. This makes the tube tend to saturate earlier. Colder bias increases the bias voltage allowing less current to pass through the tube when there is no input, giving the tube more headroom, but increasing the chance that it will reach cutoff.

To avoid gain buildup, and any potential for digital clipping, I generally make sure any block I add is "unity gain". That means I set the level of the block so the overall level doesn't change much with the block on or off. This also helps level your patches, and ensures that each block is getting an input in its sweet spot - the input level it was designed to accept. Hopefully the 2.9 level meters will help with this. But you can use your ears to get pretty close, or a level meeter in your computer, or even a dB meter in your phone.