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Posts posted by somebodyelse

  1. I used the one that came with my Helix. Think it's called an A to B cable? 

    I have a few of 'em, same as my scanner cable, MIDI keyboard cable, Faderport cable, various other PC connectable fx units, my audio interface cable... same as the one in your picture. I'm surprised you don't already use one for another piece of gear.

  2. 9 hours ago, hideout said:

    This thread reminds me of the Ideascale request I posted.  I think that an LFO that could be assigned to control and vary the delay time would be a more natural sounding chorus than what is currently available.

    They already do it.

    It's called a 'phaser'.

    THE most natural way to create chorus is to record the part a second time. In the extremely unlikely event you do it exactly the same as the first time, make a slight EQ change, a slight pitch change and add a slight delay... which is what a chorus pedal does.

    If you're using a chorus pedal and it doesn't sound "natural", try turning the knobs down. 


  3. You're basically after emulating latency...

    If you're on Reaper, take the two 'ms' numbers at the top right of the window and add em together. That's the starting point for your delay time. Otherwise, start around 20ms and adjust to taste, somewhere between 20-40ms will probably work.

    Single repeat and a mix level approaching 100%/0dB - again adjust to taste.


    Alternatively, method I've been using since the 80s is a +30 cents pitch shift at a 30% mix level.

  4. On 2/19/2021 at 5:14 PM, rd2rk said:

    FWIW - I've used the Microsoft AV and CCleaner for years without problems, but everybody's systems and surfing habits differ, so YMMV.


    No issues with MS AV, but AVG were involved in a bit of strife a few years ago when they released an AV for Windows phones. For reasons I don't understand, it turns out AV is unnecessary on Windows phone OS, which is why no one does one. Turned out the AVG app was basically data mining. AVG was never that good an AV anyway and it's owned by Avast.

    When Avast bought CCleaner a few years back, I, or more to the point, my AV, found it started installing all sorts of adware at the same time.

  5. Glad you sorted it.

    FWIW, I've been using Kaspersky Total, and Internet Security since 2007. Never let me down.

    Funnily enough, a kid on the Reaper forum posted a similar issue, just after I read this... I gave him all the above pointers... not heard back, I'd be surprised if it wasn't a similar cause.

    You clearly know yer stuff, but for the benefit of innocent bystanders, I'd steer clear of Avast and AVG and also the once reputable CCleaner.

  6. On 2/5/2021 at 8:10 PM, OmniFace said:


    "so it'll be 70Hz on the low end, too."  Not necessarily.  Having multiple speakers means more surface area.  This results in better low end response.  This is why subwoofer speakers are usually larger sizes like 15" or even 18".


    That said, the manual does state all PowerCab models have a range of 70Hz-20KHz.  So, you're correct in the end.  :)

    The 112 vs 212 presumably have their own EQ compensation curves (for multiple reasons), and the fact that they have the same response listed in the manual means they presumably account for the different bass response down to 70Hz.


    I bet the 212 has better low end response below 70Hz though, due to the doubling of surface area, but L6 doesn't provide a chart to look at.


    They also have different cabs, which will make the biggest difference overall - unless they literally did just glue two 112+s together; Alot depends in what the inside of the cab looks like, especially in a closed back cab.

    I very much doubt Yamaha/Line 6 did any research in to it and just quoted same specs to cover their a***es - nobody's gonna complain if it sounds better.

  7. Only just saw this thread.

    I was going to chime in about checking for viruses and then lecture you about free AV not being worth the money, especially Avast/AVG... 

    Sounds like you're on top of it. I'd recommend Malwarebytes for tracking down the malware. I'd be inclined to start backtracking on any free VSTs you've added recently... basically any free software you've added recently.

    Bitdefender is one of the good guys, so it's likely you've introduced the virus via software install or attachment download.

  8. 20 hours ago, PierM said:

    I dont use the Link with my two PC+ for that same reason (I could use a EQ block but DSP are full for my preset), other than because the daisy chain has few bugs that make the entire thing pretty erratic.


    Indeed, let's hope for a patch that cover both the problems. :)

    You're not getting it.

    You CANNOT EQ a digital signal stream. If you want to EQ the audio, you HAVE TO either use DSP and do it within patches OR do it AFTER the digital to audio conversion.

    ANYTHING AND EVERYTHING you do to a digital signal uses Digital Signal Processing - DSP.

    IF you're using a Powercab, you can EQ it at the Powercab side, uaing IT'S DSP and saving a little DSP at the Helix side.

    It's not a problem. It's not "patchable".

    It's like asking for dehydrated water. 

  9. My suspicion is that the Global EQ is applied 'post DA conversion', hence only being applied to analogue outs.

    You can't actually 'EQ' the digital output signal, simply because it's not an audio signal. It's a digital stream of patch information... it'd be as pointless as EQing a fax machine output. Making the digital output level higher doesn't make the audio louder, either.


    I've used the analogy in other threads... your analogue outs are like talking on your phone. The digital outs are like sending the conversation by email. No matter how hard you hit the keyboard, the writing looks the same at the other end...If you want to imply a mood, you have to include that information within the email/patch.


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  10. You're sending a digital code down a wire, NOT an audio signal. It doesn't pass through DA converters, and is strong enough to carry over several hundred feet.

    Increasing it's signal level would only make the digital signal capable of travelling further, not louder. The 'loudness' information is carried within the code, so you have to change that coded info per patch.

    I suspect the 'digital' global settiings only apply to SPDIF, in order to boost the signal level over longer cable runs, since they're unbalanced cables, low voltage and probably prone to signal degradation even over relatively short lengths. I very much doubt they actually affect the 'loudness' of the audio. The 'L6 Link' IS an AES/EBU signal.


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  11. On 4/19/2019 at 6:08 PM, PaulWilliamson said:

    ...I know its not a major issue, but its just puzzling and I wonder if its by design or just some anomaly of the firmware.


    No, it's not an anomaly.

    The difference between a digital signal and an analogue signal is like the difference between sending an email and making a phone call.



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  12. On 4/20/2019 at 4:40 AM, knoxdaz10 said:

    1. Helix signal path without cab block plugged straight into Powercab. (so i need to set the output block to Digital, because Multi will have the guitar coming out my monitors on performance

    2. Powercab using LF Raw as the amp in the room sound.

    3. Output from Powercab to DAW w/DAW set to use Powercab as sound card. ( this wont let me playback my DAW from the helix XLR, if Ableton is set to use powercab as the soundcard.)


    i really need helix for playback in Ableton, and to Record from the Powercab.

    sux because i can do this so easy in Reaper. but i just got Ableton and i wanna use it lol


    I'd just stick with Reaper.

  13. Yes and no... ebfore I relate my experience, what guitar are you using?


    I'm not a Helix user, yet - still mulling it over.

    My 'rig' is Mesa Triaxis based rack setup. I usually play a custom made ESP HSS Strat-a-like through it. Despite the gear appearance, I'm NOT a metal player, I can assure you. I roll between clean to bluesy levels of OD, I guess and as you should expect with the amount of dough spent, it sounds incredible.

    Now, I'm not getting any younger and I don't play 'the rig' at home. A few months back, I 'treated' myself to an Epiphone Dot. Yeah, it's cheap, but it plays well and I can't really justify a couple of grand for the real thing, just for home. I play it through a Vox AC4TV and it sounds pleasant enough. It's had pots and caps, picikup adjustments and improved massively. New pickups made a slight difference...

    Anyhoo, speaker went in one of 'Deep Thought's' cabs a few weeks back, so decided it was time for a change in them and bought a pair of V30s.

    Installed, in my living room, connected everything up and grabbed the nearest guitar - the Epi...

    Hells Teeth!!! That thing sounds devastating.

    Moral of the story, if your guitar sounds 'Meh', don't waste money on pickups. Get a better amp.


    Another one, FWIW, I use a 'well known amp simulation plugin' in my DAW and get a really good bass sound by feeding the DI through an 'AC30' with a 'EL34' power stage and a 4x15 cab - an approximation of an AC50, I think.

  14. Don't know.

    You need to read the respective manuals, all of which are available online.

    IF the LT has a MIDI out AND the switching outputs on the controller are the same config as the Boogie (I expect they will be), then yeah, it'll do the job.

    FWIW. I use a Voodoo MIDI controller to control a Mesa Triaxis based rack. The instruction manual for that product references the Triaxis a few times, which suggests they work towards making their gear compatible with Mesa products.


    I've read that someone is using one with a Dual Rectum Frier, so the Mesa side of things is fine. It's just down to whether the Helix LT has MIDI out.

  15. Okidoki, I'm still pondering taking the plunge in to Helix/PowerCab+ world, BUT...


    The point of the L6 link is that being a digital link, it saves the Helix and PC+ from having to convert the signal from digital to audio and back again 'in between units'. This saves processing power in both units, keeping latency low.

    The other, better reason for linking the two 'units' digitally is 'that harshness' that a lot of people complain about in digital processing and recording is usually the result of 'inter-modulation distortion' caused by AD and DA converters - a good reason to use the high cut filter on the PC+.

    Whilst I'm aware of how and what causes this distortion, I'm afraid if I try to explain it, I'll get it wrong somewhere along the line. There are plenty of explanations out there, suffice to say, the less to-ing and fro-ing between digital and audio the better.


    EDIT: Of course, everywhere I've said "digital to audio", I should have said 'digital to analogue' - it'e pnly audio after it leaves the speaker.

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