craiganderton

It's good practice with any gear to allow for the free flow of air around it. For example, with rack gear, it's common to leave a space between rack gear that generates a lot of heat.

Aside from Silverhead's helpful advice, note that many of the presets are designed to show specific techniques mentioned in the book, while others are intended as presets you can use live or in the studio. I think HX Edit is ideal for doing what you want. Back up a setlist and use it as a test bed. Drag a preset into HX Edit and play through it. If you like it, save it to a preset slot. Otherwise, move on to the next preset and try that out. After deciding which presets you want to keep, store them in a folder on your computer. Then, bring them into setlists for specific situations (e.g., 4-string bass setlist, 5-string bass setlist, cover band gigs setlist, etc.) as appropriate. Chapter 10 has 31 presets designed specifically for bass, so start there. However, I'd recommend testing out some of the guitar-oriented presets, as many of them are adaptable for bass by substituting different amps or effects. For example, Chapter 7 on Combi-Band processing is useful with bass because of how you can process the midrange separately from the bass and highs. I hope this helps! I would very much appreciate your comments on which presets you find most useful with bass, because I expect to expand the number of bass presets in version 1.4 when it comes out. I'd also be interested in what kind of bass presets you'd like to see in the future. To avoid taking up bandwidth here, please post your comments in the official Helix book support forum. Thanks!

Sorry, the forum size limits won't let me post any more uploads. But hey, the eBook that includes the graphs is only $19.95. With 459 pages and over 300 presets/files, you might find some other useful stuff in there :)

Also make sure the cable isn't a power-only cable, liked the kind designed exclusively for chargers, but also includes the data lines.

There are some rules of thumb: Moving a mic closer to the speaker increases the amount of bass. Moving it further away reduces bass, but also affects midrange. The precise midrange effect depends on the distance. Moving the mic horizontally across the speaker reduces brightness as the mic gets further away from the center. The main effect of mic angle seems to relate to phase. It think it may be less important with bass than with guitar, possibly because bass has longer wavelengths due to the lower frequencies.

I did a frequency response analysis for the Helix mics and cabs in The Big Book of Helix Tips and Tricks. Unfortunately, I've used up my allocation for image attachments in this forum, so I can't include any of the frequency response graphs. As they say, a picture is worth a thousand words, but hopefully the following descriptions included with the graphs will be helpful to you. (Helix is a great processor for bass - one of my favorite presets uses a pedal to do pitch slides. The sound is almost like a synth bass. Fun stuff!) v3.50 Bass Cabs Mics How these mics respond depends on how close they are to the speaker, because this influences the bass-boosting proximity effect. The Brute was the standard cab chosen for these measurements, so the various mic responses are influenced by the cab’s response. 40 Dynamic Although the PR 40 is often used with bass, it’s a versatile microphone that’s used in many recording scenarios. The presence lift in the upper midrange makes the PR 40 popular for vocals, as well as miking acoustic sources that need a solid bass response. 52 Dynamic The Shure Beta 52A is designed specifically for kick and bass. For emphasizing bass with a sharp attack, Shure recommends placing the mic 1" from the speaker, on-axis with the speaker cone’s center. Placing the mic further back (but still on-axis) softens the attack and reduces the bass. The 52’s frequency response resembles Shure’s quoted Beta 52A response with the mic placed about 2 ft. away from the speaker. 88 Dynamic The response curve looks like what Beyerdynamic quotes for placing the mic about 4 inches away from the speaker. Closer settings increase the proximity effect, hence the bass response. The even, extended high-frequency response is another distinguishing characteristic. D6 Dynamic This is another mic that’s clearly biased toward its intended application of miking kick drums and bass amps. The significant low-frequency boost gives a sense of fullness. The high-frequency boost provides articulation from high-frequency pick and string noises. Furthermore, the scooped mids reduce the potential for boominess and mud. This is a specialized mic that does what it’s supposed to do. Bass Cabs 1x12 Epicenter This final group of cabs is for bass, so a horn usually adds upper midrange emphasis. Without this, the bass would have a hard time sounding defined in a mix. The peak in this cab is more subtle than most, and the response is smooth. 1x15 Ampeg B-15 The B-15 has a reputation for a deep, full bass sound with plenty of articulation. This curve shows why: there’s a significant bass peak, midrange scoop to make room for other instruments, and a broad peak centered around 3 kHz. 2x15 Brute From the midrange on down, the response is relatively flat. This is likely due to the unusual, individually tuned, multi-port system. The upper-midrange peak is from the horn, which contributes the high frequencies needed to emphasize “string zing” and articulation. 4x10 Garden This splits the difference somewhat between the previous two bass cabs. The Garden has more of a scoop than the Brute, and a slightly reduced low end compared to the B-15. 4x10 Ampeg Pro 4x10 cabs tend to have more similarities than differences, and 4x10 Ampeg Pro and 4x10 Garden are no exception. However, the Ampeg Pro has a narrower upper-midrange peak, and a peak at 10 kHz for more “air.” 8x10 SVT AV This is the only Helix bass cab whose physical counterpart doesn’t include a horn. As a result, this one’s all about the bass, with a midrange scoop and a peak around 4 kHz.

I really enjoyed everything you wrote...except this. Best thing I ever did for myself was getting this set of IEMs with built-in ambiance mics. They're expen$ive, but to have "high-fidelity earplugs with a volume control" is nirvana for my ears. I even wear the IEMs to concerts and movie theaters where the sound is too loud. Full disclosure: I endorse these IEMs but am not compensated for doing so. I just believe strongly that musicians should protect their hearing. Sorry for the detour, back to why Helix is so effing wonderful!

Is there a reason you don't want to use the Twin Harmony? If you just want a major harmony, specify the Key, Major, and the amount of Shift (for a major harmony, it would be a 3rd). The voice quality isn't great, but if you slather some reverb and/or chorus to thicken it, and trim the high frequencies to remove the Mickey Mouse effect, it can give decent vocal support. The caveat is that you need to mix it as low as possible behind the main voice, consistent with hearing at least a little of it. Just don't expect miracles

You need to update Helix Native as a separate download from the firmware for hardware units. Go to downloads, specify Helix Native as the software, choose your operating system, then click on Go. After updating, check the About window on your Helix Native. Mine says 3.61, with a copyright date of 2023.

Thanks for the comment! It's inspired me to come up with what might be a useful addition to v1.4. In v1.3, there's a section in the chapter "The Overachieving Audio Interface" about using HX Stomp as a pseudo-plugin for DAWs. With Windows, that means your only audio interface can be Helix (Macs don't have that limitation). However, many Mac and Windows DAWs include a way to interface with external hardware. Your hardware Helix then appears in your DAW just like it was a Helix Native plugin. So, the idea would be to expand the existing section to explain how DAWs handle external hardware, and then include some cool presets for drums, vocals, etc. Helix is outstanding with many more audio sources than just guitar - some of the ways I'm using Helix with drum buses and drum loops can't be done with any other software I know of. It can also do superb effects with background vocals, strings, etc. Even though I have a ton of plugins, there's a preset I use on my voice that has become my go-to for vocals. I'm curious if anyone would find this kind of material useful. Comments?

Question to the community: How interested are you in using Helix, and especially Helix Native, on instruments like vocals, drums, strings, synth, etc.? I'm using Native a lot in my studio projects, mostly because while it does the "bread and butter" presets well, it does more than traditional plugins. So far I haven't added this kind of information to the Helix book, but I'm considering it...and maybe even releasing a set of studio-oriented presets in the Marketplace. It seems like an unexplored area that could use more attention.

Good point, I guess the solution depends on whether the FOH mixer is more concerned about peak levels because of distortion issues, or average (perceived) levels so that when a sound appears, there doesn't have to be a massive change in the fader setting. (This assumes the rest of the band plays at a consistent level, and it's mostly the guitar player's levels that concern the FOH mixer).

Here's an excerpt from my eBook, The Big Book of Helix Tips and Tricks. It's somewhat of a deep dive, so I hope you find it helpful. How to Level Presets People have different attitudes about how (or even whether) to set presets to the same output level. There are four complications when trying to set consistent levels: A sound’s perceived level can be different from its measured level. A brighter sound may measure as softer, but be perceived as louder because it has energy where our ears are most sensitive. You want some sounds, like solos, to have a louder perceived level than others. With live performance, different presets will have different perceived levels depending on room acoustics, the size of the audience, and the music you’re playing. Presets that sound good in a home studio over monitors may not work well for live performance. When using presets onstage, the only way to guarantee setting the right levels is to adjust them while playing live, in context. Regardless, having a consistent, baseline level speeds up the tweaking process. Here’s an analogy. When adjusting a pickup’s pole pieces, I screw them all in halfway. Then if a string needs to be louder or softer, I can adjust the pole pieces as needed. If they’d all been screwed out, I couldn’t make them louder. If they’d been screwed all the way in, I couldn’t make them softer. It’s easier to tweak preset levels if they’re already close to what you want. The following is intended for those who are familiar with recording, editing, or mastering audio. If your head explodes, move on to the "About Gain Staging" section in this chapter. A Partial Solution, Borrowed from Mastering Engineers A measurement protocol called LUFS (Loudness Unit Full Scale) measures perceived loudness, not absolute loudness. The origin story (every superhero has an origin story, right?) is that the European Broadcast Union (EBU) had enough of mastering engineers making CDs as loud as possible, in their quest to win “the loudness wars.” LUFS measurements allow streaming services like YouTube, Spotify, Apple Music, and others to adjust the volume of various songs to the same perceived level. So, you don’t have to change the volume for every song in a playlist—that Belgian hardcore techno cut from 1998 sounds like it’s the same level as Billie Eilish. The system isn’t perfect, but it’s better than dealing with constant level variations. LUFS meters measure perceived levels. Some DAWs include LUFS meters. Several third-party plug-in meters are available, like the Waves WLM Plus Loudness Meter, or the Youlean Loudness Meter (the company offers a free basic version). The goal with leveling presets is for their outputs to have the same LUFS reading. This is not a panacea! You will almost certainly need to tweak output levels for specific performance situations and musical material. However, having a standard output level makes tweaking easier, because you’ve established a standard. Presets may need to be either louder or softer than the standard. Setting the Output Level If you don’t care about whether the preset has the same perceived level as a dry guitar (you don’t need to), adjust the output to whatever sounds right. However, I find comparing the processed sound to the bypassed sound is a useful baseline. Creating consistent preset levels is much easier to do with a computer and Helix Native. Import the presets into Helix Native, tweak them, then transfer them back to your stage devices. Here’s the process: 1. Set the Helix Native input and output levels to 0.0, and don’t touch them. You’ll make any needed input or output level adjustments in the preset itself. 2. Record a 15-30 second or so clip of bypassed guitar playing chords, without any major pauses, and another clip with 15-30 seconds of single notes, also without major pauses. For bass presets, record some bass lines. 3. Insert an LUFS meter after Helix Native. Depending on the preset, loop the chord or single note clip at least two or three times with Helix bypassed. Check the LUFS reading. 4. After enabling Helix, reset the LUFS meter, and again play the same loop at least two or three times. Adjust levels within the preset to hit the same target LUFS reading. 5. Whenever you change the preset level, reset the LUFS meter reading before playing back audio into Helix. 6. Use your ears to do any final output level edits, based on the musical context. Tip: Most LUFS meters measure the instantaneous LUFS level as well as an average level over time. If you play the loop through a few times, the average level will settle to a final value, whether bypassed or through an effect. This is the reading you want to use, not the instantaneous one. Note that this isn’t an exact science. The object is for your presets to have a standard, baseline level. That way, when you get to the gig, massive tweaking probably won’t be needed.

If the block is before the distortion, you'll send a clean signal through the send output (with the level set by the Send parameter). The same clean signal at the Send block input will exit the block and proceed to the rest of the Helix chain.

It's the kind of sound that indeed, is more obvious in isolation. But it's also the kind of effect that you don't notice initially. When you get rid of the fizz and then re-introduce it, you'll wonder why you didn't notice it before. In the video I linked to, if you don't want to take the two-and-half minutes to check out the whole thing, skip ahead to 1:55. The sound includes two notches to remove fizz. Compare that to the original sound, which for comparison, plays after the sound with the notches. I'm sure you'll hear the difference. When you play the original sound for most people, the first reaction is "I don't hear anything wrong." But when you add the notches, and then remove them, then it's easy to hear what I'm talking about. I certainly agree the new cabs and mic placements greatly reduce fizzing, but it can still show up with various settings and be identifiable in the context of a mix.

A steep notch filter at the output, tuned to the fizz frequency, does it for me. I wrote an article (with audio examples in a YouTube video) 13 years ago for Harmony Central. I recommend checking it out to solve the majority of fizz problems. (FWIW because of this article, Avid released an Expansion Pack for the Eleven Rack that included a parametric notch filter at the output. And, a company I consulted to recently on some amp sims held off the product introduction to notch out the fizz frequencies in their amps after I pointed them out. So it's not just in my imagination!)

I have found one possible exception to the "don't put effects between amp and cab." Here's an excerpt from the section on optimizing pitch-shifter quality in version 1.3 of my Helix book. It seems counter-intuitive, but try it. I was surprised, and don't really have an explanation of why this seems to be the case. With pitch-shifters designed for single notes, playing more than one note at a time can give a hot mess of a sound. Playing cleanly improves the sound quality. By definition, pitch shifters that generate harmonies create more than one note at a time. For these, effects chain placement is crucial: • If an effects chain includes effects that generate distortion (including amps), it’s usually best to insert the pitch shifter after distortion. However, if the sound quality is acceptable when playing chords without a pitch shifter, inserting the pitch shifter before distortion may work. • Inserting the pitch shifter between an amp and cab often sounds better than inserting after an Amp+Cab. This is true for the non-polyphonic pitch processors. • With a clean effects chain, most of the time it’s best to insert the pitch shifter at the chain’s beginning, because then it gets the cleanest possible signal.

No problem! The way I figure it, I have to keep the books updated anyway for those who are just discovering Helix. So, making the update available to previous buyers doesn't require work other than what I'd do anyway. Besides, I don't feel it's right to charge people for a new book if much of the material was already in previous versions. I'm sure there were be a v1.4 at some point, so suggestions are always welcome! There are still many Helix aspects I want to cover that don't relate to an update. A good example of going back and fleshing out older material is the analysis in v1.3 of what the tone control does in the three-parameter distortions.

Thanks, I've made the change. I'll wait a couple weeks to see if anything else shows up, and then do a 1.3.1 hotfix. Unfortunately, my awesome proofreader was in and out of the hospital from March until a few days ago. She's a better proofreader than I am for my books, because she can see them with fresh eyes. Please let me know if you think the info on the 3-control distortions was useful. If people like it, I'm considering doing a deep dive on some of the other distortions for v1.4.

If you own any version of the The Huge Book of Helix Tips & Tricks, it's time to go grab your free update to version 1.3! Here's what's new: 489 pages (30 more pages than version v1.2.1, 121 more pages than version v1.1) Frequency response curves and analysis of the 9 new cabs introduced in Helix v3.60, and revised frequency response curves for the cabs introduced in v3.50 Coverage of the new Grammatico GSG, Elmsley, and Agua Sledge amps Coverage of the new Dark Dove and Triple Rotary effects blocks All-new tone analysis of the 13 three-parameter distortion blocks (this doesn't relate to 3.60, it's just something I found fascinating) And as the software folks say, "additional bug fixes and enhancements" :) To obtain the revised version, download the book again from the Sweetwater site at the link above, using the code you used to download your previous version. (If you didn't save the email with the code, it's in your Sweetwater account history.) Remember there's a dedicated support forum for all my Sweetwater books on the Musicplayer.com website. I welcome suggestions for future revisions. Finally, I've mentioned before that when a book goes to a new edition that represents a significant change (like software going from version 1.X to 2.X), owners of previous editions will be able to get the new edition for half-price. However, I don't anticipate this happening for quite some time.

As an addition to the great information presented by Lorenzo, if you're interested in an elaboration of what he says above, the following is an excerpt from my Max Your Mix! eBook. I hope you find it useful! Here are the main attributes of several pan laws: • Most common. Most DAWs default to a -3 dB equal power pan law. With stereo, a mono channel panned from left to right will have a constant perceived level. If a DAW doesn’t default to this pan law, it will be available as an option. • Best for collapsing to mono. With stereo mixes that will collapse to mono (e.g., for broadcast or playing over a mono sound system), most engineers choose a -6 dB pan law. This provides constant gain. This means that a mono channel panning from left to right in the original stereo mix stays at the same perceived level when played back in mono. With other pan laws, mono audio will sound louder as it passes through center in the original stereo mix. • SSL console standard. Solid State Logic consoles use a -4.5 dB pan law. SSL was the first company to include compression on every channel. With a -4.5 pan law, a channel compressor panned to center reacts differently compared to other pan laws. There’s a subtle depth to the mix. You can consider this pan law like a preset that provides a specific effect. • “Up front” sonic character. The -2.5 dB pan law gives a somewhat punchier sonic character. This is also like a preset that provides a specific effect. • Old-school linear gain change. This allows for a center-channel buildup when panning a signal from left to right. It’s preferred for extracting the mid signal from a stereo track for mid-side processing. As laws go, pan laws are not applied universally. Different programs can default to different pan laws. This can become a problem when moving a project from one host DAW to another. Unless the selected panning laws match, choosing identical level and panpot settings may not produce the same sonic result. Perhaps when someone thinks a particular DAW sounds “punchier” than another, the “punchy” DAW might boost the level when signals are panned hard left or right. The “unpunchy” one might use the law that drops the center level instead. For example, suppose you move a Cakewalk project to Cubase. The project will sound softer, because even with the same basic pan law, Cubase drops the center to maintain equal power, while Cakewalk raises the left and right sides. Conversely, if you move a Cubase project to Cakewalk, you might have to deal with distortion issues because signals panned hard left and hard right will now have a higher level.

One more thing...when I was gigging on guitar with DJs and electronic bands in Europe at the turn of the century, because I wasn't singing, I could be in front of the PA. Not only could I hear the guitar perfectly because I was going directly into the PA, I could hear it in relation to all the other instruments. I'd like to think this helped me contribute to a better overall mix. The guy running FOH just left me alone, nothing needed to be done because I knew EXACTLY what the audience was hearing. It was what I wanted to hear, too :) It's all good...the right answer depends on the question that's being asked! If you're playing with a hard rock act at a festival, for 4,000 Germans whacked out on ecstasy, or at a local club with a rock band, I think the question and corresponding answer differ.

Did a festival gig once going through a Bose L1. Amp in a room, or FRFR? I dunno, but what was really cool was with that column speaker, I could place the guitar parallel to the speakers and get awesome feedback :) It felt like I had more physical interaction with that amp than I'd had for a long time with an actual guitar amp. But to be fair, it also had a bass bin that was blasting air, like an amp. Frankly, if only guitar amps existed, I'd play through guitar amps. If only Helix and FRFR existed, they I'd play through that. Maybe I'm just not that much of a "feel" guy. I played through guitar amps throughout my high school and college years but when I went pro, I went direct through keyboard amps and never looked back. Because I straddled the world of studio/live, it was the perfect solution because it worked for both. So, when modelers came out and would let me take that to the next level, I was already primed. This isn't to say that the feel of a guitar in a room isn't cool. But bringing the sound of the studio to the stage can be pretty darn cool, too...just in a different way. I've experienced both, and I truly believe it's a matter of personal preference.

I hear ya. FWIW, I ditched guitar amps completely in 1968 (to be fair, tubes had really cratered in quality anyway) and switched to keyboard amps. They were the best you could in terms for FRFR at the time. My stage setup was two 100W RMI amps. Amazingly enough, they could even fill arenas. My goal was to get the sound I wanted before hitting an input jack. As a result, the sound was the same whether playing live or in the studio. That made life a lot easier, especially when I started gigging with DJs in Europe in the 2000s - I could just patch right into the PA system, and all I needed was an AdrenaLinn, POD, and Vocoder (fed by drums). I could fit it all in a carry-on bag for a transatlantic flight. Does it sound like an amp in a room? To me, no. To the audience, yes. And they're the ones buying the tickets :)

Perhaps there's a fundamental issue - FRFR is designed for a flat response, cabs are designed for a NOT flat response. Cabs have a variety of anomalies and boosts/cuts. The engineers who design FRFRs have total control over the electronic circuits that drive the speakers. They do everything they can to remove any anomalies, in order to obtain a flat response. It might be that you're missing the "warts" in FRFR. Sure, the Helix emulates cabs, but if you're talking about physical wood vibrating in an acoustic space, software can't do that - even if you feed it with an impulse of physical wood vibrating in an acoustic space.