craiganderton

I've started on the 3.5 update by doing frequency response graphs of the new cabs. Then I want to show how the various mic positions affect the sound of selected cabs. For example, Line 6 models the proximity effect on directional mics, which comes into play when you move the mic closer to the speaker. It's most noticeable with the ribbons. However - if there are specific areas you want to see covered, drop by the forum! There is much about Helix that the book doesn't cover, like doing automation with DAWs, setting up footswitches, that sort of thing. I'll be prioritizing the new features, but if there's a sense that people really need some other features covered. I'll see what I can do. Also this probably goes without saying, but since I don't have a firm ETA yet, if you don't have the book you don't have to wait for the update to appear. You can get the current version, and grab the free update when it's ready. Thanks everyone for your interest and support!!! Doing these books is a bit of a labor of love, so it's good to know they're helpful to Helix fans.

I analyzed all the mic models (as of the previous OS) for my Helix book. If you have it, check Chapter 6. The U87 has a more extended high-frequency response,which is why it's so popular for vocalists in the real world. It's also good for dry guitar. The U67 doesn't go quite as high, and has a bit of a bass bump. Here are the images from the book (top U67, bottom U87): So, if you want a sound more like a U87, use EQ on the U67 to add a gentle treble boost starting around 2 - 3 kHz (shelf EQ would be a good place to start), and lower the bass response around 100 Hz. But also remember that the response for either mic depends on whether they modeled the cardioid, omni, or figure-8 mode. I'm assuming cardioid based on the frequency response graphs, but that's a guess. Many engineers feel the U67 has a more "natural" sound than the U87, so it's often used for instruments as opposed to the U87. One of the major differences is that the U87 handles higher SPLs and the U67 has a lower output impedance, but that doesn't matter in our virtual world. Hope this helps...

I'm glad they took are of you, good to hear. But I forgot to mention a crucial point: Save the download code they gave you! You'll need it to download the free updates (you use the same code to get the free "point" updates). Normally this is stored in your account, but if you sign in as a guest, it won't get stored on the Sweetwater site.

Thanks for your interest! In case you missed it, I posted this in the "Big Book" thread: There's a dedicated support forum for all my Sweetwater Publishing books in the forum section of musicplayer.com. I just posted in the Helix thread today. Please note this isn't only about me posting stuff about when to expect an update!! Feel free to post suggestions about what you want to see covered in future updates, any problems you encounter with any of the tips, any typoze you find, anything that's not clear, and of course, anything you particularly like so I'll keep going in that direction.

I didn't want to use up forum bandwidth for something that could appear to be self-serving, so there's a dedicated support forum for all my Sweetwater Publishing books in the forum section of musicplayer.com. I just posted in the Helix thread today. Please note this isn't only about me posting stuff about when to expect an update!! Feel free to post suggestions about what you want to see covered in future updates, any problems you encounter with any of the tips, any typoze you find, anything that's not clear, and of course, anything you particularly like so I'll keep going in that direction. The thread has quite a few views, but people aren't taking advantage of the two-way nature. Remember, I'm writing these books for you, so the more I know about what you want, the better :)

Most people know not use a USB hub, but I've been told by people who know a lot more than me that using USB ports close to the computer's power supply (e.g., rear panel) are better than the front panel ones. True? I have no idea. But I don't have update problems, so maybe it is. I also use short cables that are spec'ed for USB Gen 3 data streams.

For me, a mix of amps is ideal. I like the classics too, but currently I probably have more projects using the Cartographer and Ventoux than other amps. As to Sunn amps, they were a big deal back in the 60s. They were big and loud, in an era when guitar players were trying to fill stadiums with a Fender Twin. It was a favorite among bass players, but guitarists liked them too. They were also relatively clean.

I received the following from Sweetwater: The customer should be able to go through the checkout process without having to create an account at the very least. We have occasional international orders for software, which the e-books would mimic in delivery behavior, so things should work just fine. If they continue to have an issue, I would have them reach out via the email form at this link: https://www.sweetwater.com/about/contact/. It’s monitored constantly, and someone should be in touch within the day. So I think the takeaway here is the contact info. Sweetwater is very good about support. There is not yet a mechanism in place to notify customers when a new version is out. I do know it's in their "to do" list, but I'm not aware of an ETA. Meanwhile, if you want to find out about current versions, future updates, have suggestions for what you would like to see in future versions (hint, hint!), etc., there's a dedicated support forum for all my Sweetwater books at Musicplayer.com. I monitor it at least every other day. One of the threads lists all books and their current versions. Hope this helps! (FYI my next book, The Musician's Audio Handbook, will be available starting Friday.)

I sent a note to Sweetwater asking for clarification about this. If it really isn't possible, I'll come up with some kind of solution. I appreciate your interest, thank you.

Not quite the same thing, but I made a box to find out how different impedances affect the guitar sound. It's basically just a rotary switch and resistors. Here's a link to the article and schematic.

When checking out sag, I found that it varies tremendously depending on the amp. These oscilloscope shots are from my Helix eBook:

I use the Oscilloscope plug-in in Studio One. For example, here's an image of the Ampeg Opto Comp's attack from my Helix book (version 1.1). The attack on the left is with Compression set to 5.0 on a 100 Hz sine wave, while the one on the right uses a setting of 10.0. Note how the 10.0 setting clamps down faster on the attack, and reduces the sustain level. The relatively slow attack gives a “smooth” punch, and prevents any buzzing on low-frequency bass notes.

I understand where you're coming from. Look at how many amps and effects have stood the test of time. It's ironic that some people want Helix to model amps and effects that haven't changed in decades, yet are dissatisfied with Helix's "progress." It's good to remember all that matters is the emotional impact on the audience, and that's up to the player, not the gear. No one ever came up to me after a concert and said "Wow, loved the sound of that multi-core chip!"

Interesting discussion, for sure. My main use for EQ cabs is when I want two cabs in HX Stomp, and there's only enough DSP for one. So EQ makes it possible to do parallel paths. Of course everything is subjective and everyone has their own desires of what an amp should sound like. I feel the sound consists of what comes out of the cab, what happens in a room, and your position in the room. More specifically, if you're playing guitar in a room, you're almost certainly moving around. So the frequencies and phase shifts your ear experiences are constantly shifting. In the CA-X amps I designed for Sonar, a big part of the sound was extremely subtle chorusing - not as an effect, but just to introduce constantly shifting, sublimiinal phase shifts and response anomalies that move over time. I think that's an important time element that's often overlooked, and is why some people think the sound of an amp sim is "missing something" if they're listening over headphones, or to studio monitors while sitting down, in the studio.

If a sim has a cab sound you’d love to use with Helix, it’s not only possible—it’s easy. The process requires a DAW, and sending an impulse (a short audio file) into the virtual cab you want to capture. After loading the impulse into a DAW’s audio track, insert the cab whose impulse response you want to capture as an effect for that track. Bypass all amps and effects—you want only the cab to process the impulse. Render the track, and export it as a WAV file. Then you can load this into an IR block. That’s the short form, The Big Book of Helix Tips and Tricks eBook goes into the details. The download for the book also includes impulse files in mono and stereo at various sample rates that you can use with your DAW. The process for different DAWs is similar, but there are specific examples for Studio One, Pro Tools, and Ableton Live. So...you don't have to buy me a coffee or a beer, just buy the book :)

A good example is sag. The physical power amp might not have any significant amount, but the power amp in the Helix might.

It just occurred to me...maybe the kick drum pedal could be hooked up to a switch that would duplicate the sustain footswitch function mentioned above. Then you wouldn't need to do any conversion of kick notes to controllers.

Then it's doable, but you need to be able to map the MIDI note driving the kick to a continuous controller. An easy way to test this out is to turn off Helix hardware compatibility in Preferences. Then, use a keyboard's sustain footswitch (controller 64) as a controller to trigger the compressor. My favorite for this application is the Ampeg Opto Comp. Trigger the Compress parameter, with Min Compress at 0.0 and Max Compress at 10.0. Every time you hit the footswitch momentarily, you get the pumping effect. Adjust Release to taste. (If you want to keep hardware compatibility enabled, choose a switch controller other than 64 that Helix doesn't reserve as a global controller.) I haven't tried triggering from note data, though. Ableton Live can use Max routing to trigger CCs from notes, but I don't know what can be done in Logic. However, MIDI Solutions' hardware Event Processor can map just about any MIDI event to any other MIDI event. You can download the programming software without buying the unit to see what it does. Or just contact the company and ask them if it will convert note data to continuous controllers, they've always been helpful in the past. Once you map the kick drum note to a continuous switch controller and map the controller to the compressor, you should be good to go. Again, I do this with a footswitch. So, I know that you can achieve that sound, but I haven't gone through the process of converting a MIDI kick drum note to a switch controller. Hope this helps! In any case your question helped me, because it was a reminder to include the "footswitch pump" tip in a future revision of my Helix book :)

Excellent question. I got an HX Stomp when writing the book because I wanted as much info as possible to apply to HX Stomp and Helix LT/floor/etc. Of course, the detailed descriptions of the amps, effects, etc. are equally applicable to both. Of the 12 Chapters, one is exclusive to Helix. It covers 3- and 4-band multiband processing, which is not possible (mostly) to do with HX Stomp (there is a pseudo-3-band workaround for HX Stomp where one path processes the midrange, and the other, the lows and highs). However, there's a lot of material in the book on how to save DSP power. Many preamps can substitute for amps if you know a few tricks, and use less DSP. Two different amps can share one cab and still generate a stereo image from the Delay and Reverb, like this Of course, you may have to compromise on which amps and effects you choose, but you might be surprised at how many DSP tradeoffs you can make, and still have solid-sounding presets. Also, I cover using EQ-based cabs, which save quite a bit of DSP power. And many presets don't involve dual amps at all. One of my favorite clean sounds layers a Nashville-tuned sound with dry guitar: If you have specific questions, I've opened a dedicated support forum for the book, where people can get support if they have questions about the tips, as well as give suggestions about what they' like to see in future revisions. A lot of great suggestions for the 1.1 version came from readers. I have to say that as an author, interacting with readers to shape a new version, instead of just putting a book out there, is a very cool experience. If you like, ask me to post the table contents in the support forum, and I'll post it so you can see exactly what's in the book. There's also a promo video about the book, although it was made before the version 1.1 release. Hope this helps!

The linked post shows the settings - it's cranked up to +20, but the ideal value for your setup depends on the input level going into the EQ, and the input level/drive setting on your amp. I don't hit inputs all that hard, I need the extra gain. Unfortunately, there's no "one-size-fits all" setting...which come to think of it, pretty much applies to anything involving amp sims :)

Yes. It reverses the phase.

I thought some of you might find this helpful...boosting EQ around 3 kHz will make an overdriven amp more responsive to leads. The screenshot below shows four identical riffs. The first is played starting on low E, the second an octave above, another octave above that (i.e., starting on the open first string), and finally, the octave starting on the first string, twelfth fret. The lower image shows the same file, but processed with a major boost around 3 kHz. The level of the lowest frequency’s riff is essentially unaffected, the second one is boosted somewhat, but the two higher riffs have considerably more level. This makes for a more even, sustained sound with solos played on the higher frets. If you want more info, I wrote this up originally about Guitar Rig for a Native Instruments blog post.

You're correct that a de-esser uses compression to clamp down on the highs. It's basically dynamic equalization. However, most commercial de-essers use a notch filter, to zero in on the offending frequency. Clamping down on the entire high end could (but not necessarily!) cause the audio to sound dull. With Helix, using a parametric stage set to a notch that's tuned to the ess frequency may be more effective. The problem is you lose those frequencies regardless of whether or not there's an ess sound, but the frequencies above it remain intact, which may be good enough. In the studio, I use Helix Native quite a bit as a vocal processor. It has "guitar" functionality that other vocal processors don't have.

At least with a reverse delay, you can have delayed sound only. The mental gymnastics required to play 1 bar earlier what you want to hear 1 bar later isn't easy, but it's not too horrible if there aren't a lot of chord changes.

Totally agree, after I got Helix I sold my Mutron. I'll have to try your post-cab idea, I haven't done that but it sounds promising. Another option is split early in the chain to a path with compressor and octave divider. For Wes Montgomery-type jazz octaves, delay the dry sound by about 20 ms to emulate hitting the higher string later than the lower string.