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  1. So, I downloaded Reaper and loaded Native in there, and everything is fine. It's only happening in Reason. So I did a search, and came over these: So apparently this is a thing. Anyone got any suggestions on how I could figure this out? Is this a bug with Native, or a bug with Reason? Should I submit this as a bug? I dunno. For now, I'll just use Reaper, but Reason is my main DAW.
  2. I've had Native for a few years, messed around with it a little at the beginning, but haven't touched it since. I do know I've had this problem a few months ago when I tried to set it up, and I just gave up due to other constraints. But so now I'm trying to set it up again. Everything is working just fine except one thing. Whenever I change a parameter inside the Native VST window, like literally every time I change the mix or decay rate, or even just touch an on/off toggle on any block, or literally if i just click my mouse buttons anywhere inside the VST window, the snapshot changes to a random other snapshot. I'm not running an elaborate MIDI setup where a bunch of messages are being sent and received. It's just Native, running inside Reason 10, with a guitar plugged straight into my Focusrite, and using the mouse for editing. No HX hardware is connected, nor any other MIDI controller or anything else for that matter. No other audio or MIDI software is running on the PC. The only things connected to the computer are the USB audio interface, the HDMI, the ethernet, the keyboard, the mouse, and a wireless receiver for the mouse. Nothing I have going on should be interacting with the VST. If I press the slider on a parameter and hold the mouse button, the snapshot doesn't change until I let go. Just like how footswitches can be set to momentary/latching/on press/on release, it seems like this is happening "on release" after changing literally anything inside the VST window. Needless to say this is making it near impossible to hear the changes I'm making. I'm guessing there's something I'm not understanding up in all this, I just can't imagine this is how the software is supposed to act. This is really weird behaviour, and I can't for the life of me figure out what the thing is about. Is there some hidden mouse keybind that I'm triggering? How do I remove all such things? I won't be using this live, I don't even need the snapshots. It's just for simple practice at this point. But it's completely unusable as is.
  3. What exactly isn't working for you? You're getting no sound? Lol silly question, but like, I don't get the problem.
  4. They said they wanted to use the new poly engine to make some new reverbs that they really wanted to get into 3.0, but didn't make it - paraphrasing. So it feels like a safe bet that 3.1 might have some new reverbs.
  5. Lol. You're not exactly wrong, but wow that's a pessimistic outlook. Let me present you with a more positive, and more truthful perspective: It's not that nothing's ever good enough. It's that what's good for some people isn't good for other people. It's not like it's the same people that's complaining about everything. I complain about things once in a while, but for every complaint I utter, I spend something like a hundred or a thousand times more time not complaining. I'm not sure why you're only registering complaints, but you should rethink your outlook. It doesn't sound healthy. And it's wrong, because a lot of people are super happy all the time. You have fallen prey to one of the logical fallacies, I'm afraid. Shake it loose.
  6. "QOSP QOSP" could be used to describe the sound of someone chewing cheetos puffs.
  7. This is true. That said, no, those blocks would not tie up the first DSP on the Helix. In my current preset I have these blocks on paths 1A and 1B (they are all mono versions): Para EQ - Simple Pitch - Legacy Phaser - Legacy Analog Flanger - KWB od - Teemah! - Scream 808 - Wah - Bubble Vibrato - 10 band EQ - Tremolo - Volume For the heck of it, I just tried adding a stereo Chorus, stereo Trinity Chorus, and stereo 70s Chorus. That's when some blocks started graying out for me. On the first DSP. So about 6-10 stereo blocks depending on what types of blocks, and estimately twice that amount of mono blocks is possible on one DSP chip.
  8. Didn't know where else to report it, but I think I discovered a bug. Whenever I try to overwrite an IR already on the PC+ with a new IR, the old IR gets overwritten but the old name still appears in the IR list. If I try to rename the file, the new name re-appears, and if I hit "enter" it goes back to showing the old name again. If I remove a letter, hit enter, +then restore the letter and hit enter again, the new name finally gets stored and appears in the list. And I'm not sure if this is a bug, but I am running two PC+ in stereo through L6 link. If I connect either of them to my computer through USB, I only get one of the sound channels through to the PC. Which means I can't use either of them to record the full signal they are being sent. Not a huge deal, but since you can't use two USB interfaces to record at once that seems like something that should (could?) get fixed.
  9. DT/PC+ amp control without MIDI is on the feature list.
  10. That only affects the XLR output on the Powercabs, not the sound in the speaker. If it does for you, I don't know the explanation, but the conceptual idea is: when you play through the Powercab with speaker models used, it is supposed to sound like an actual amp or cabinet in the room. So no mic or preamp in the signal. But if you want to record that sound, instead of using a regular mic and a mic preamp, which is the go-to solution the powercab simulates those on the XLR output. It's part of the "amp in the room, all in one stage rig" thing they're going for. Helix straight into PC+ is all you need. That is how it is intended and advertised to work. As for your question about levels, I just figured I'd chime in. You may all be right, or wrong for that matter, I don't know, but I haven't seen anyone talk about why there is a volume difference. Bear with me here, but so if you imagine the internal sound signal inside the Helix, one can easily imagine that internally, it is at its "natural" volume level. When that signal is fed into an output, different outputs have different properties. 1/4" unbalanced line level is louder than XLR balanced line level. Don't mistake that though, if you have an amp with an unbalanced 1/4" & a balanced XLR output, and an active speaker with an unbalanced 1/4" input and a balanced XLR input, so long as you go from balanced to balanced and unbalanced to unbalanced, there is no difference in volume. But if you feed a balanced XLR output into an unbalanced 1/4" input, the unbalanced input will likely clip. It will be louder "than intended" or however I should put it. You will be above unity gain. You could say if you were landing a rocket, you're way out to sea. That is another thing no one mentioned. Unity gain. For the inputs and outputs to be matched and for the units to be gain staged so that there is no artificial boosting of the volume to get you were you want to be. You don't want that. It's not a matter of taste, it's a matter of what is correct and what is wrong. If you (not just OP, anyone reading) care about tone but don't know about unity gain, you should wrap your head around that before you do anything else. Especially if you got a complex rig. So when you use digital, with the volume control at 100%, you are sending the full signal that is inside the Helix to the outputs. If that's lower than before, that suggests to me that before, you were actually feeding it a signal that was "too loud". You were possibly slightly overloading the Powercab input. Considering it's not an amp, but an FRFR speaker, I can only imagine that that distorted your signal (but possibly so little that you wouldn't notice it unless you did actual A/B tests with recorded clips in headphones). I could be wrong, but it sounds like your volume isn't too low now, it was too high before. What you are now getting is the entire internal signal, with its headroom. Maybe what is actually the case is, because you were feeding the Powercab a signal that was too loud for the input, you've dialed in your patches accordingly. Now that you are at unity gain, maybe you can increase the channel volume on the amp without clipping, where you would earlier clip. You should look into that. Or connect an XLR into the PC output and record to your computer. If your signal is peaking, then I'm wrong, but I'm pretty sure if you do that, you'll have loads of headroom to go on. That's not a bad thing btw, but if things aren't as loud as you want (percieved loudness) that also means there's something you can do about it. I'm curious though, are you actually unable to get the PC to be as loud as you want when you turn it all the way up, or was it more the difference from before that was bugging you, leading you to think something wasn't right? In my rig I have two Powercabs. I run a Kemper in an FX loop on the Helix and some pedals in the other loops. There's a lot of cables and I've gain staged every unit. I run my computer sound through the Helix so when I watch Youtube videos on evenings I have my PCs' volume controls set to 30% and I still have to lower the Windows volume to between 12-20% so as to not keep my neighbours up. What I mean to say is, that volume is louder than regular living room TV volume, at 15% of 30% (so what, 4,5% of max volume). I am physically afraid to go past 50% because when I get close I can feel my ears start to hurt. Like, gunshot loud hurt.
  11. It's not that I didn't think it wouldn't work, or couldn't have tried. IF (BIG if) you know that either the transmitter or the receiver is transformer coupled and the interconnect distance is short then a simple resistor divider will match the impedances and change the level as shown in Fig. 1. This is the AES3id recommended network for creating a 12-dB pad (4:1 voltage divider) and converting the AES3 110 ohm balanced output impedance into 75 ohms for driving the AES3id input. Therefore an average output level of 4 volts will be reduced to 1 volt. Since this exceeds the max allowed for S/PDIF, use the values shown in parenthesis to create an 18-dB pad (8:1 voltage divider) producing 0.5 volt output for the same 4 volts input. (Other average AES3 output voltages require different resistors -- consult AES3id for value graph). Transformers make the best passive impedance matchers, plus provide the benefits of ground isolation, high-frequency rejection, DC blocking and short-circuit protection. Impedance matching is easily handled by selection of the appropriate turns ratio (1.21:1 for 110-ohm to 75-ohm -- it's the square of the turns ratio for impedance) and careful attention to winding details allow wideband high-frequency transformers (you need ~12.5 MHz). The best and safest converter includes the transformer. If you are not sure about the transformer isolation of the equipment interfacing, use a store-bought impedance matching transformer that comes complete with connectors and a separate resistor voltage divider network as shown in Fig. 2. This T-network is a 75-ohm:75-ohm bi-directional attenuator. The attenuation is a little less than that of Fig. 1 for the same (assumed) 4-volt AES3 input because the transformer reduces the voltage level by a factor of 1.21:1 (down to 3.3 volts) as well as matching the impedances. If you're still reading, the point is, the box takes care of a number of signal integrity issues that I would have to learn and understand a bunch of stuff i'll never use for anything but this single conversion in my rig to overcome. It cost the same as about two mogami cables, I have about 10 of those in my rig. I didn't want "works", I wanted "optimal solution". :P It has active electronics, so it kinda works like a line isolator with respects to ground. I haven't yet done any latency tests but S/PDIF and AES have the same kind of data code for the sound. The difference is header data. So there's nothing being converted. Maybe header data is being stripped. It makes no sense that there's any latency induced by the design. I can't tell a difference before or after, but presumably there's a couple ms less lag in my signal. There, at the very least, is no more lag than there used to be. It is a known fact that AD/DA causes a bit of hissing noise. So there's a tiny amount less hiss in my setup. I'm not sure I even have a noise floor any more.
  12. RCA S/PDIF and XLR AES cables have different resistances. I was assured by people more knowledgable than me that that was sub-optimal. So I got this box finally. The power adaptor doesn't work in the EU, but I was able to get a universal wall wart at a regular all supply store and that worked just fine. Eventually I'll get a better one, like maybe the iFi iPower. I falsely assumed the 9V DC input would accept guitar pedalboard power supply cables, but it didn't. It's a smaller type of plug. So it works just as advertised. Kemper S/PDIF out goes into the box and an AES cable goes into the Helix AES/EBU input. The Helix picks up the signal fine and everything works as expected. One less AD/DA round in my rig now. Too bad the Helix can't run both S/PDIF and AES at once or I could eliminate one more. I just realized, since I'm already using my 1/4" main Left output to feed the Kemper, I now have two extra FX loops available in the Helix with this box. That's another reason to shell out the 100 I didn't even consider.
  13. Co-sign, hoping there's something in the pipeline for the Powercab+. I'm using two Powercabs in stereo through L6 link and when I connect either one of them to the computer through USB I can only record the channel going to that speaker and not both channels. I think someone affiliated with Line 6 said that's a bug, so hoping that'll get fixed too.
  14. I have the Helix rack version. I just checked, you're right, it's not an L6 link input, it's an AES/EBU input. I can't use the Helix S/PDIF input because in the global settings, you have to set "Digital" to either "AES / EBU" or "S/PDIF" and one disables the other. I have two Powecab Plus' that I run from the L6 link output (which even gets expanded upon in the next firmware patch) so that's why I have to use the AES / EBU input. I've done some reading, and AES / EBU and S/PDIF may be cross-compatible or it may not. Some times it is, some times it isn't. There's the sound data and some header data, and depending on the header data, the receiving unit may or may not be able to interpret the signal. Also, the resistance is different in S/PDIF and AES / EBU cables which seems to make a difference if I am to trust people more experienced with these kinds of crossovers than me. I think a custom made AES / SPDIF cable is an untidy solution, and like I said, research seems to suggest it could end up not even working. I'm not sure there actually is a digital-digital conversion with this box, I think the box just does something to the header data. I don't really exactly know what it does, but I know it's meant for interfacing exactly like the way I'm trying to do. Different units with different I/Os. I may be a little off, maybe you don't call it header data, I didn't graduate in this, but that's the jist of what I found out. I can interface ANY unit with only S/PDIF outs to my Helix. Or my Helix to ANY unit that only has S/PDIF inputs, while using my Powercabs. That's the worth of the box. If it ends up causing a mere 1ms lower total latency in my rig that'll be an added bonus, and I know it'll have an effect on the tiny amount of hiss I do get with the Kemper noise reduction set to off. I am already using standard 1/4". That's what I want to not do, if possible.
  15. I'm looking at this: I'm wondering if anyone's tried to use this with the L6 link input on the Helix. Does it work? I want to feed my Helix an S/PDIF signal from a Kemper. I've read about S/PDIF to AES/EBU and understand that some times it works and some times it doesn't, and it's kinda down to the implementation of the standards in a given device. I'm just trying to see if anyone already tried it before I go ahead and do so myself. I'm trying to bring the number of AD/DA conversions in my rig down.
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