jonandtice

I think having a volume knob on the Helix is counterproductive. Mine is assigned to only control the 1/4" out but since there are settings for instrument/line level I don't actually ever move the knob. Seems to me like the volume knob just encourages bad habits (turning monitoring volume up after sound check, not properly volume balancing presets, etc.).

On 5/2/2021 at 3:36 PM, phil_m said:

Even if you think you did, it means you didn’t... There is no other reason. Macs sometimes seem to grab the old installation files when you think you're installing the new one. So clear out any old HX Edit dmg packages prior to updating.

 

For the love of God, don’t post another thread about it or ask it again!

Quote

 

Quote

How do I update to 3.10?

Did you happen to update HX Edit to 3.01 (released on Feb 2, 2021)? If so, just connect Helix/HX to your computer via USB and launch HX Edit. The software will walk you through the entire procedure, including updating HX Edit 3.01 to 3.10, and backing everything up to your computer.

 

We weren't supposed to have to though. HX Edit is supposed to update itself now. Didn't work for me though, using Windows. I did update it manually and got it working before I posted though.

It sounds like what happens to me when my DAW is monitoring the Helix USB signal to the same monitors that the  Helix 1/4" outputs are going into. I end up with a duplicate signal that sounds bigger because it is louder and the DAW signal is imperceptibly delayed compared to the other one.

I think we've failed to answer your original question which was what makes a standalone power amp different from a guitar amp power amp. Typically guitar amp power amps are tube power amps. That may or may not be a good thing depending on how you are using your Helix. I think solid state power amps are common in bass amps.

I haven't used one myself but the Tech 21 Power Engine Deuce Deluxe is what I would get if I was playing both guitar and bass with Helix and wanted a more traditional cab-like amp-in-the-room thing. If you want an actual guitar cabinet/speaker that works for bass and guitar, Quilter has been putting a Celestion BN12-300 (which is a bass speaker) in their Mach 2 Combo 12HD, which is a guitar amp. So a power amp with a March 2 HD Extension cabinet or a bass cabinet with a similar speaker might be an option.

I got this and it works

https://www.sweetwater.com/store/detail/XRF110--hosa-xrf110

Just to revisit what has been said in this thread before:

Technically, you should use S/PDIF connectors and cable meant for digital signals but for short lengths of cable, RCA and instrument cable work. The AES/EBU digital signal is technically different than what the S/PDIF is expecting to receive but they are similar enough that they do work together in most cases.

So your issue is probably that the connector isn't wired correctly, or that the connector + your RCA cable has too much signal loss for the digital signal.

I dial stuff in at home on studio monitors and IEM with backing tracks that are similar to what the live sound will be. I kind of have a sense now for how it will translate to the live sound. You will learn after a few tries how to adjust.

My personal opinion, if you get the Helix and studio monitors for home and IEM for live, you will have a better enjoyment balance between practice and live performance. The two will be closer together than what you are experiencing now. Even if they aren't exactly the same in both environments they will be more similar than different compared to your current situation.

Your other option would be to use a reactive loadbox such as the Two Notes Torpedo Captor. If you really like your current amp, this is what you should do.

1 hour ago, yeatzee said:

 

I've heard good things so far! Haven't actually tried it myself, just updated to do the comparison video then had to leave town but I'll probably start playing with it tonight :)

Looking forward to a video on it from you!

1 hour ago, PierM said:

 

Line6 software doesnt use procedural updates (base install/Core + updated libraries), so each update it's just a brand new download of a brand new installer, which needs previous .exe version to quit. Im pretty sure you still were on 3.01 when you tried first time.

I wasn't. I checked Help -> About HX Edit... It said 3.10. I rebooted just to make sure it wasn't using old libraries.

@yeatzee I feel like the new Dynamic Hall reverb in 3.10 replicates your sound exactly (or whatever variation of it I had). Would you concur? This is exciting!

My settings are:

Decay: 7.5sec

Predelay: 100ms

Room Size: 20m

Diffusion: 85%

Damping: 10.0 kHz

Mix: 100% (could be like a 40% mix in series and not in parallel anymore but I still like running in parallel)

Motion: 2.0

Low Freq: 100 Hz

Low Gain: 0.0 dB

Low Cut: 120 Hz

High Cut: 8.0 kHz

I had HX Edit 3.01. It told me there was an update so I clicked. It updated both HX Edit and Helix to 3.10. However, HX Edit wasn't showing me the new blocks even though HX Edit was saying it was 3.10. I reinstalled 3.10 manually and then it worked.

Sometimes the 4CM cables can create a ground loop and you need to isolate one of the cables but it sounds like you have it narrowed down to the USB. I have the same USB problem but it isn't really that bad until I have something running that makes my graphics card work hard. So I think in my case, the graphics card or the PC power supply is leaking power when there is high demand from the graphics card. Also, it's worse when my guitar pickups are facing the PC. The closer I am, the worse the noise.

14 hours ago, rvroberts said:

I'm surprised you don't know AC30s don't have FX loops,.....

Except for the handwired ones, all the modern production AC30s have FX loops. I have an AC30C2 and wouldn't have bought it without an FX loop. I run it in 4CM sometimes and it's great.

22 hours ago, fasteddiesanchez said:

Thanx for the response Phil_m. So my next question would be about this chain:

 

stereo effect1 -> amp -> stereo effect2 -> dual cab block.

 

 

 

 

 

 

 

 

If you need something stereo happening before the amp and you want to preserve the stereo separation, you could do this:

stereo effect1 -> Split Y A:L100-> amp -> mono effect2 -> mono cab block -> Merge A:L100 
                          B:R100-> amp -> mono effect2 -> mono cab block ->       B:R100

However, I usually like to place effects before or after amp/cab in which case you could do this place the a dual cab after the merge point and place effect2 as stereo after that.

46 minutes ago, CraigGT said:

Are you sure about clock source to internal rd2rk?

Whenever I've used SPDiF in the past (admittedly not with my Scarlett) it needed to be taking sync from the SPDiF

You are correct the sync should be S/PDIF but mine works as internal too. I just get some audio clicks once in a while. Also audio RCA work fine as long as they aren't too long.

2 hours ago, jonton2 said:

Thank you so much. So the reason I have the Grand Canyon Looper in the Send/Return block is so the looper has access to anything I put in front of it with the stomp, yet still will get picked up in my DAW. Thanks for your response. I appreciate your help as I am pretty much a noob still with the stomp and settings.  

 

Let me ask you this.  Is it possible for the right output to go to Peavey / Left goes to the DV Mark - yes playing both at same time but neither get CAB/AMP/IR like you stated, yet still have the AMP/CAB/IR available for the DAW only? I guess that is what I am really looking for. So in summary - no AMP/CAB/IR to either AMP, but still available to DAW/Headphones only.  You rock ! ! ! 

OK I missed the part where you had a looper in your FX loop, so your FX loop makes sense now. The main problem you have right now is that the bottom path should be set to USB 1/2 and your FX loop should be before the split if you want the looper output in your DAW.

Your signal path should be:

[Input] -> [Effects Blocks] -> [FX Loop] -> [Split] -> [Output1]
                                                    -> [Amp Block] -> [IR Block] -> [Output2]

Output1 would be set to Main L/R like you have it and those would go to the DV and the Peavey as you describe.

Output2 should be set to USB 1/2 and if you adjust the setting I gave you above (recommendation #4) your headphones will give you what you want.

Your setup is confusing to me. I'm going to assume here that you are using the Stomp as your audio interface.

First of all, do you use the DV and the Peavey at the same time? If not, then just make two separate presets to simplify things.

Second, in your screenshot I don't think the FX loop block is serving any purpose for you. You are going to your DV preamp, coming back into the Stomp and then right back out to the DV return? Where is Main L/R going? Just eliminate that FX block because you need its send for the Peavey.

Third, do you really want to apply the IR to the Peavy? Doesn't it already have a guitar speaker? You normally wouldn't want to do this but there are exceptions.

My recommendations:

1. Delete the FX Loop block and your Main L would go to your DV preamp input. No need to return to the Stomp unless you want to put effects after the preamp. Do you want to move that mod block before the path split?
2. Between the AMP block and the IR block, insert a Send L/R block. That will go to your Peavey.
3. Select the output block on the bottom path and change that to USB 1/2.
4. In the global settings set the Phones Monitor to USB 1/2.

Now you should have your No-amp, no-ir signal going to your DV (top path); Your AMP, no-ir signal going to your Peavey (send block from bottom path); your AMP+IR signal going to your headphones and DAW via USB 1/2. I think that is what you are looking for but, as I said above, if you are not playing through your DV and Peavey at the same time, separate presets would be better.

I haven't used one but the BluGuitar Amp1 uses a "nanotube" as seems like it somewhat gets the tube thing?

15 minutes ago, PierM said:

It's same format, but protocols are different (there is an extra bit, CH STAT, carrying info in a different way), and also there are impedance differences (as well as different Vp-p).

 

Id grab a converter to avoid the problems you are getting.

To clarify, if I set the clock source on the audio interface to S/PDIF, I don't have any problems. I don't think that "problem" would go away with a converter because the clock still has to be synced between the two devices and Helix LT isn't able to receive clock from other devices.

In case anyone is wondering about connecting the Helix LT through the AES/EBU digital out I'll share my thread on this topic over at TGP.

https://www.thegearpage.net/board/index.php?threads/helix-lt-aes-ebu-to-interface-s-pdif.2227695/

Original post:

Quote

I have a Helix LT and a Focusrite Scarlett 8i6 USB audio interface. I had TS cables connecting the Helix to the interface but then I thought, why not keep everything digital if I can? So I got an XLR to RCA cable, put the XLR end in the Helix AES/EBU out, put the RCA end in the Scarlett S/PDIF in, and it worked!

Then I started hearing clicks in the audio every couple of minutes so I'm guessing either the 10' XLR cable is too long for digital signals or it's a sync problem (I'm guessing from reading about S/PDIF in the Focusrite manual). So I set the sync source for the Focusrite to "S/PDIF" and, as fare as I can tell, no more clicks.

Should this have worked? I feel like I'm cheating. I had initially thought I would have to buy a box like this to connect the two. I'm guessing the Focusrite can't act as the clock source for the Helix (box or no box) since it is a one-way digital connection? Is this correct and are there any downsides to having the Helix be the master clock?

I'm really happy to be connecting this way since it removes one set of digital-to-analog conversions. The Scarlett 8i6 is able to direct monitor the S/PDIF signal even when my computer is off which I also wasn't expecting. Next, I plan on doing a comparison of the digital signal and the analog signal through interface so I can see how much time I'm wasting to achieve an imperceptible benefit.

Someone will probably ask:

Quote

Why not just use the Helix as your audio interface?

My response:

Quote

Since my Helix LT goes in and out of the house about once a week, it isn't always at the computer. Sometimes I just plug straight in to the audio interface, use Helix Native, and don't bother taking the Helix out of the bag. I use the computer for a lot more than just playing guitar so it is more convenient and it keeps things more consistent to have the studio monitors permanently pulgged in to the audio interface.

If I was to do any serious recording where I wanted to do reamping I would either:

1. Use the audio interface and Helix Native.
2. Use the Helix LT as the audio interface.

If Windows/ASIO allowed multiple audio interfaces to be used by the same DAW, that would be the dream. This is possible with Linux/Jack.

My understanding is that what is in the the 5.7 kernel is basically a hack to get the Helix to work as a class compliant device.  If Line 6 would fix the Helix, instead of pretending it's fully class compliant, it would work everywhere.

You can do it several ways.

I have my Helix 1/4" outputs going to my audio interface and the studio monitors are plugged into the audio interface. However, when the Helix is on I use it as the ASIO audio interface and the actual audio interface is free to be used as the system sound device. Or if the Helix isn't on, the audio interface becomes the ASIO device (assuming you're using Windows).

What you describe will work but you will limit yourself to mono output and a single path of effect blocks.  Instead, I would suggest using a send block as the first block in your chain. But if the Helix is on and you want to record wet and dry at the same time, just use the Helix as the ASIO device. You can still monitor it through the audio interface.

Phil_m is right on. Sometimes, if you don't have space to play, you shouldn't be playing. In my experience, if you are playing with a great pianist, just give up and hang back because most church pianists do not know how to play with a full modern worship band. Unless your worship leader understands arrangement and space and is willing to address it nothing will change. And if the pianist is the worship leader, well, good luck.

15 minutes ago, chasingMango said:

Can you verify that your input signal strength is the same?  What audio interface are you using for Native?

I'm using a Lexicon Alpha, using the instrument input and the trim at 12 o'clock.  Pretty sure I'm getting the same signal strength at the input.  The Native and LT presets sound identical and I perceive the volume to be the same once I increase the Native output +18dB.

I've been playing around with Helix Native after having the LT for a while.  I have several presets that I have dialed in on the LT to peak at around -9dB (I shoot for between -12dB and -6dB) when putting out a Line Level signal with the volume knob at 100%.  I imported the same presets into Helix native and the output from native is around -24 dB using the same guitar.  It's almost like the output from Native is behaving like my LT with 1/4" outputs set at instrument level.  Thoughts?

My Helix LT does not work as a class compliant USB audio interface, however I haven't tried to use it since 2.8 firmware came out and I haven't updated to Ubuntu 19.10 yet.  Maybe I'll try again tonight.