dragonfet

Could you attach your preset to a post? I would suspect there's a level problem hidden somewhere.

I would try to set the Send and Return levels to line level (individually) and see if that helps. You may need to fiddle with the levels: If you set the Helix Send to Line instead of Instrument leveel, you may have to reduce the Send volume a little. TRS cables would only help when using symmetric connections. I don't know if the Ebtech does that, but a more intricate approach would be to use two passive DI boxes: Helix (TR cable) --> DI (XLR cable) --> DI (TR Cable) --> Amp. The TR cables should be short while the XLRs can be long. With stereo DI boxes like the Palmer PAN 04 you could run two connections fully symmetrical. The disasvantage is obviously the need for two DIs.

You are absolutely correct, thanks for the suggestion. I just did the test again with just the Amp-Cabsim path on 2.30 and there is a difference between the "new" and the "old" recording, again at -84.2 dB. I deduct that there is no strict corellation to the firmware version change, so it looks even more like there is no change at all. I have edited my original post.

I seriously doubt that. Given that IRs are probably recorded with a samplerate of 96 kHz or even more, the only question is what resamples them to the 48 kHz the Helix uses. If you want that to be done by HXedit, you should feed it the original files (or anything not 48 kHz), if you want to use whatever the IR manufacturer uses you should get the 48 kHz file from them. But in any case, the difference is either neglible or not even there as resampling isn't magic.

You can also use the volume of the output block, but other than going through the patches manually and adjusting them with your ears I see no way of doing that. Maybe a special view in HXedit wherer you have all snapshots / presets in a list with volume knobs next to them, but even then you'd have to do it by ear. A peakmeter doesn't tell you how loud something sounds, so you'd need an intricate algorithm to do that automatically. And it reminded me of those wonderful "Auto" switches on compressors, gates and nowadays probably entire channelstrips. Don't they always sound so much better than adjusting them yourself...?

Let's head back to the original topic. I hadn't updated until today because my Helix Floor was in the bandroom. So I decided it might be a good idea to answer the question "Did something change with IR and/or cabsim sound between firmware versions 2.21 and 2.30?". TL;DR: Yes, but the impact is minimal.There is no measurable difference. [Edit, see below] My setup is as follows: - Modified Ibanez PIB1 goes wirelessly into the Helix AUX input - DAW: Reaper - 96 kHz / 24 Bit throughout the process I used the bridge pickup in coilsplit mode which gives a thin, guitar-like sound that emphasizes the highs. I recorded a single take with low and high notes in Reaper, then reamped it through the Helix with two signal paths: 1. Industrial Fuzz into an IR: 2048 point Mesa 2x12 IR 2. German Ubersonic into a cabsim: 4x12 XXL V30, Mic 57 Cabsim Both sound harsh and too broad for any useful application, but provide a good amount of high frequencies for the test, which was as follows: 1. Record both paths on Firmware 2.21 2. Backup, Update 3. Restore Backup, Reboot to let Helix rebuild the patches 4. Record both paths on Firmware 2.30 5. Invert phase on the 2.21 tracks 6. Render each pair of tracks into a single WAVE file respectively Since the phase is inverted, the result would be "nothing" (a WAVE file filled with zeroes) if there is no difference between firmware 2.21 and 2.30. This is true for the Fuzz-IR path, but the Amp-Cabsim one is not reproduced exactly the same. Reaper normalizes the difference file to +84.2 dB, which means the actual change is happening at -84.2 dB. This is absolutely inaudible - aside from those people who hear the solder joints in their cables. Edit: Following a suggestion by hefonthefjords, I compared two Amp-Cabsim-recordings of the same source take in 2.30. The result is again a slight difference at -84.2 dB, which means that this difference is not related to the firmware version. There might be even smaller smaller differences (below 84.2 dB) that are hidden in the natural variations of the Helix modeling, which can't be detected by this test. But then again: These would be impossible to have an audible impact. Keep in mind: I was testing a single preset. It is not said that the difference is nil / -84.2 dB for all other possible combinations. Results may vary.

A peakmeter for every block in the chain.

I'm from Europe and I use Cordial midrange cables. In my opinion expensive cables don't make sense as there is no audible difference between mid and highrange. Lifetime warranties are included in the higher price, but even with a replacement it doesn't pay off. I can recommend making your own cables though, especially for rack cabling, as it allows you to get just the right length for each one.

Update: This has now been confirmed as a bug and might be fixed in a future release.

Most wireless receivers should be able output line level. I strongly recommend to check the receiver's manual for that. 10 kOhm is a great input for wireless, I use that for my Shure GLXD6 receiver. No colouration is going to happen with the AUX input, it's only difference are lower level and impedance compared to the GUITAR in. Also keep in mind that a lower impedance usually means less noise. It's not that relevant with short signal paths and line level, but never assume that higher input impedance is better.

I have not tested that, but you can put a Looper and all the Send/Returns into a patch as long as you have enough space. Also the Looper takes a bit of processing power, s you can't add it to patches with a huge amount of effects.

Keep in mind that it could not just be the coax, but anything else that connects to a different earth / ground pane. Sadly ground loop removal is a very tedious procedure. Aside from that, I'd like to leave an opinion: Power conditioners will not help here. An overvoltage / transient protection is very useful, even more important is something that tells you whether you have good earth etc. Some also contain hamonics filter that make sure your attached devices will only get a more or less pure 50/60 Hz sine and no "dirt", e.g. from fluorescent lamps. I already doubt the Helix would be affected by that though as it has a switch mode power supply - there is almost no way that harmonics from the power line can couple into the DC side of it. So how should it help against hum that not only occurs much deeper down in the circuit, but also in conjunction with another device? Plus, if the hum is at 50/60 Hz, it's not a harmonic, so no filter could ever get that out of the Helix' or the amp head's power. I'm curious to hear whether a power conditioner helped anyone get rid of a ground loop. Until then, I'd like to be the counterweight for those who keep recommending them for all kinds of issues, from noise over hum to tube eating Marshalls.

I last used the 4CM with my Boss GT-10B and an Ashdown head, so I can't say much about Marshalls and such. But a few general things: 1. Does setting the Send on the Helix to Line level cause the Return of the head to be overdriven? If not, or if you can adjust the Retun gain, I would try setting the Send to Line level as that should provide a bettee Signal-to-Noise ratio. 2. How long are your cables? Asymmetric connections like guitar amp send/returns will always pick up a lot of noise, and it gets worse over distance. 3. If you have two passive DI boxes, try to use them to form a symmetric (XLR) connection between the Helix and the Amp. Passive DIs also work in reverse, so you can use them on either end to turn 1/4" into XLR and vice versa.

I noticed it when playing and even more so when I was testing different pickup wirings. I recorded short dry samples of different configurations, then used the Helix in Reamp routing to test them with different amps. That's where it became obvious. I'll open up a ticket now as you suggested. Though I'm still curious if others have the same issue.

This occurs with any sound being fed into the SV Beast, including normal bass signals. I just included Method B so that other users can try to recreate the symptom without firing up their DAW, but if you think it's unsuitable I'll remove it. Method A should still show the problem.

I'm referring to my thread Bug Confirmation: SV Beast Sag Clicking. It would be helpful if someone can confirm or disconfirm the symptom I'm experiencing. Symptom: Both the SV Beast Nrm and Brt models exhibit a clicking noise after sound was processed if the "Sag" parameter is set to a value of 0.6 or greater. Steps to reproduce: 1. Create a patch with just an SV Beast Amp in it. Set the "Sag" to a value equal or greater than 0.6, the other controls don't seem to have an influence. Set the input block to something quiet, e.g. USB. Method A: 2. Connect the Helix to a DAW and use a routing with sound only going in and out via USB (Reamping). 3. Monitor the Helix' main output (Channels 1&2), e.g. by arming a track for recording. 4. Play anything for any amount of time, then stop playback. Method B: 2. Set the "Hum" value to anything that produces an audible sound, then move it rapidly to 0. For both methods, there should be a crackling noise following that sounds like static discharge going through a lowpass filter. For "Sag" settings of 0.5 or less, I could not reproduce the symptom.

I want to ask if someone can verify that the following happens on their Helix as it does on mine before I file a bug report. I searched for this before creating this topic, if it's a dupe I'm excusing myself beforehand. I use Firmware 2.21 . I recommend using a DAW or method B to make sure there's no additional noises coming from pickups, cables etc. Symptom: Both the SV Beast Nrm and Brt models exhibit a clicking noise after sound was processed if the "Sag" parameter is set to a value of 0.6 or greater. Steps to reproduce: 1. Create a patch with just an SV Beast Amp in it. Set the "Sag" to a value equal or greater than 0.6, the other controls don't seem to have an influence. Set the input block to something quiet, e.g. USB. Method A: 2. Connect the Helix to a DAW and use a routing with sound only going in and out via USB (Reamping). 3. Monitor the Helix' main output (Channels 1&2), e.g. by arming a track for recording. 4. Play anything for any amount of time, then stop playback. Method B: 2. Set the "Hum" value to anything that produces an audible sound, then move it rapidly to 0. For both methods, there should be a crackling noise following that sounds like static discharge going through a lowpass filter. For "Sag" settings of 0.5 or less, I could not reproduce the symptom. If someone can confirm this, I will file a bug report, otherwise I'll scratch my head. With a confused face.

That does not work and you seem to use the wrong word here. You can "mix" them, but then they are merged into a single mono channel, which doesn't seem to be your intention. XLR has three pins, but it's a symmetric connection, so all three pins are used for one mono signal. If you want stereo, you need to use two XLRs, one hard panned left and one hard panned right.

Can you clarify? You have two channels in the DAW, one for Left (Amp 1) and one for Right (Amp 2) I take it. These are mono channels and XLR plugs are usually mono too.

I recommend using a send or the right channel 1/4" if you don't need that. The send has the advantage that you can route it to a useful spot, which would be before any effects that might upset the tuner, e.g. chorus or phaser. You can specifically insert a pure Send Block that has no return. I would refrain from using the headphone output. Even if you use a stereo cable, the tuner input is likely mono and it will short the right channel on the Helix' headphone amplifier. It is most likely rated to tolerate that, but it's avoidable. If you have to use it, I recommend using a split cable ("insert cable") so you can leave the right channel unplugged. The same restriction applies to it as with the 1/4" Line Outs: a pitch modulating effect in the signal chain might confuse the tuner.

Hello Jeff, as a happy Helix owner I can contribute the following: 1. Ease of use: It took me a bit to fully understand the concept of the Helix' UI, but once you're in it, it's amazingly easy. Getting the patches to work well takes some time, e.g. you can't have two snapshots with very different gain structures in the blocks as that will create an audible pop when switching. That's probably what the other two posts meant with "tech savvy" and "know how things work". It's a non-solveable problem though: The more (virtual) knobs and switches you have, the more possibilities you have to screw everything up. I work a lot with different distortion sounds and heavy effects, so I sit a lot of home with the unit to build new patches and snapshots. When I come to the band room, I usually adjust them a little to work with my gig setup and at gig volumes, that's something you should prepare for if you want to make tones at home. If you have a setup as simple as you described (EQ, Chorus etc. all in series), I'd think it's just a matter of adding those blocks, configuring them to your taste and calling it a day. Maybe create Snapshots if you have eight or less different bypass states or if you want different settings for them. I'm very picky with distortion, so it took me a while to find the right ones to use for the right occassions, but there is a lot to choose from. Also the modeling seems very accurate; it reacts very sensitive to dynamics if you want it to. If you meant "live" literally: There is a short moment of silence when changing patches, so you should try to use snapshots or stompbox mode instead. I have had zero problems during gigs, and I doubt the LT will perform any worse. I set the unit down, connect power, bass, an XLR cable for the FOH and the 1/4" for a monitor if necessary, that's it. 2. Pitchshifting: I just tested it. It seems to work normal to good on the upper four strings of my Ibanez 5-String, but the low string has tracking issues on all pitch related effects. The first two frets create an added low frequency interference that varies with picking intensity, the low E string seems to take longer to track. It doesn't matter how much you downtune, since the problem likely is with the note tracking itself and not in the processing that follows to change the pitch, add harmonics etc. - I cannot recommend doing that. Sorry if this post focuses too much on my personal setup, but I hope it helps your decision.