FUNDAMENTALS for Getting GOOD Tones from Guitar Plugins like Neural DSP and Helix Native
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- เผยแพร่เมื่อ 28 ต.ค. 2022
- For my Presets visit: johnnathancordy.gumroad.com/ Guitar Plugins are, I think, a HUGE part of the future landscape of guitar tone. However, there are some FUNDAMENTAL tips and pointers that I've learned from making mistakes over the years, that are REALLY important to ensure that you're getting a) Good tones into the Plugin and b) Good results out of it....
Have you got other tips for getting decent tones and results from plugins?
/ johnnathancordy get my backing tracks here
www.buymeacoffee.com/johnnath... - if you like what I do and
only if you wanted, you could buy me a coffee!
Get my PADs bundle here: www.paypal.com/cgi-bin/webscr... (this folder will grow, and I'd encourage you to ask for different keys/songs and stuff that might suit this type of preset?)
I've decided to make it possible to grab both my Helix/HX Stomp bundles (the expression bundle with freeze presets has always been separate) together for £5 - www.paypal.com/cgi-bin/webscr... - I will then email you a link to both bundles!
Try my general patches for Helix or HX Stomp in this bundle for £3 using this link - I will then send out the patches! www.paypal.com/cgi-bin/webscr...
You can get my EXPRESSION patches in this bundle using this link - I will then send out the patches! www.paypal.com/cgi-bin/webscr...
Try my patches for Pod GO for £3 using this link - I will then send out the patches! www.paypal.com/cgi-bin/webscr...
/ johnnathancordy get my backing tracks here
www.buymeacoffee.com/johnnath... - if you like what I do and
only if you wanted, you could buy me a coffee!
2:04 Plugins are the future BUT
3:24 IMPEDANCE - what you need to know
5:25 No "Hi-Z" no problem
6:48 LATENCY - the hardest problem to fix
9:59 Get SIGNAL levels right BEFORE recording - เพลง
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One point on Latency as well - Drivers can actually make a huge difference to this - so if you haven't checked that you're using the latest Drivers for your Audio Interface - this might be worth investigating to get better performance?
with that said, other device drivers (network, wifi, graphics card etc...) all do affect overall DPC latency of a system (and therefore also cpu cycle buffer load), so it's always not a bad idea to keep all those up-to-date and fully functional ;)
One thing that tripped me up when I started experimenting with guitar plugins was noise from the guitar pickups being too close to the computer. This was another thing I was surprised the plugin companies don't mention. A computer puts out an electromagnetic field (EMF), and when I'm sitting right next to mine so that I can use my mouse to twiddle settings, my guitar picks up a lot of noise from the PC. With high gain settings, this is incredibly noisy. I just needed to back away from the PC and then maybe rotate around a bit to get much lower noise, but that didn't occur to me right away.
Always good advice about latency/drivers. Once I’m done with all my tracking, I will often set the buffer to maximum. This gives my the most options as I add plug-ins to my tracks and/or master buss. You don’t need low latency when mixing.
Another reason for the hx stomp. It works really great as an audio interface. It even has the possibility to record your dry signal alongside the processed signal.
Most modellers will do this, but I agree that the Stomp is extremely handy as an interface. Takes up very little desk space too.
So I realized the wet/d.i. thing immediately too.
does it have monitoring? Meaning can it output my other instruments from the daw?
@@tusharjamwal yes, via the headphone out
@@tusharjamwal Yes
The Waves PRS Amp plugin actually tells you on startup that you should put a guitar pedal with a buffer in the signal chain.
The last tip is GOLD. Last month was when I realised that the Green light on my interface indicates healthy levels, NOT clipping haha. Good stuff John!
a thing to consider. there's latency in real amps as well. the further away you are from it, the longer it takes the sound to hit your ears. if you're playing with headphones or sitting directly in front of your monitors then SOME latency won't hurt. I find 64 with monitors and 128 with headphones to be fine.
To clarify latency comes from Turning a signal from analog to digital and processing that. Latency shows itself as a delay, a side effect of latency. You won't have latency with an analog gear but you will have delay depending on how far away you are from it simply because the speed of sound (analog sound from your amp) is drastically slower than the speed of light (analog signal going to your amp). John I'm confident you know this I'm simply putting the information here for people still learning.
Mathematically, for every meter (~3 feet) you are from your analog amp you're adding ~3 Ms of delay from when your amp makes noise to when you hear it. so if you're at 5ms of latency in a VST and it's in your headphones, in theory it could be FASTER than a regular amp that you're several feet away from. Sorry if this is too much info.
Not too much info at all. It's good info.
This is really important, especially considering all latencies in hardware, like wireless units for guitar. I thought that my Xvive latency is pretty high, until I realized that I'll have the same experience standing 2 meters from the amp :P if this is not a problem, neither will that :)
I’ve been struggling with this for a while. Thanks!
I was hoping you'd go into your process for dialing in a preset. Great content per usual. Cheers.
Very helpful lesson for me. Thank you. Haven't gotten into plugins yet and want to start.
For anyone whose computer is struggling to handle those small buffer sizes for low latency, it might be worth monitoring the guitar separately. A cheap passive DI box would allow you to split the signal - link/jack output goes to a guitar amp (or digital multi-fx etc) just for your monitoring, while the XLR output goes to your audio interface. This also means you're sorted if your interface lacks a Hi-Z input since the DI box goes to a mic input.
I was giving up Helix Native because I could not find a way to make it sound good through my Yamaha mixer/interface. The latency issue and the always on dry signal made the Native frustrating…I didn’t want to use the Helix Floor all the time for logistic reasons and everything. Well, I decided to get a Focusrite 3g and WOW! That was exactly what I needed. An interface with very little to no latency, pure wet signal and the focus on the plug-in, which was what I wanted. Great plug-in, I even use it to rehearse at home because voices sound amazing too…great video. Thanks!
Love every intro I’ve ever heard you play 👊🏻
Pro tip for gain staging is to check with a palm mute as bass frequencies take up more headroom, therefore you may set it to not clip by hitting a bar chord hard, but then the moment you palm mute it goes to red.
Solid advice - thanks Cordy
Thank you for this really helpful information!
I didn't know about impedance with bypassed pedal into interface. Such MASSIVE difference. Thanks
Amazing playing
Hi John, I'm glad you are discussing the software side of things and would like to see more. You should really try to get a hold of a IKMultimedia AXE I/O USB device. It has a few features geared toward guitarists that are unique. It includes a variable impedence knob for dialing in the sound of your guitar. I think a followup video would be worthwhile. Great content.
The AXE I/O comes with not so decent latency figures, especially given it's a dedicated guitar interface. Others are doing much better.
It's been 2 years I've been into music production and never have I heard about the low/high impedance inputs. Thank you!
If you are thinking about buying a pc for music production you may want to focus on CPU more, especially look for single core/single thread performance as this will more impact the latency than the other stuff.
Great video! Latency drove me crazy for quite a while and I finally landed on an interface with “direct monitoring” which removes the latency problem but makes using an amp sim tricky. I usually record a modeled signal from a strymon iridium then sometimes record an additional dry direct track in case I want to add an amp sim when mixing later. Thanks for the tips!
Yes!!!
Yeah, direct monitoring removes the problem of latency, but you also end up hearing a dry guitar track that makes it not fun to play.
Personally, I find that "low latency mode" in Logic allows you to still hear the signal out of your amp sim but without any latency. It does this by turning off a bunch of routing and other plugins in the session that are causing latency. Often the root of latency is not the amp sim. It's a reverb on an aux or a limiter on your master bus.
great playing
When you pulled up the behringer with the 6.5mm adapter I was looking at mine cause it is exactly the same!!
Thanks for the the video.. Super thankful for you man .. if you have time could you do a quick video on stackable presets… i’ve tried to figure this out on my own and I’m just kind of confused… ( Hx Stomp )
John I’m so relieved you have this video in your arsenal. I only today bought Archetype: Rabea and so far I can’t get it to sound anything other than sh*t. But I’m relieved because I have the same Behringer UMC22 interface so I’m 99% certain my hardware is not the problem - just the idiot controlling it all!
Gold material right here, I now know all these things now but it took me a lot of trial and error to get it right, had a lot of output and impedance issues with my HX stomp when I first bought it, to the point I stopped using it, but eventually I figured it out.
To clarify something about amp sim input controls, they're not there to adjust the signal level (almost always they are level-compensating) but to emulate pickup output.
If you're doing proper level calibration for each guitar using your audio interface's input gain control, all of the D.I. tracks should be of similar level regardless of the actual output of the pickups.
That's where the amp sims' Input control comes in to re-adjust those levels. So, turn the Input control down to emulate low-output pickups and up to emulate high-output pickups. What that does is change how hard the signal hits the amp sim's preamp.
I"m having issues with this exact thing. I get crazy string noise unless I roll my plugin input back to at least -2.0. I have to adjust every plugin preset input to get a decent sound. Is this an issue with my interface gain setting? I try to set to -15 or so, but I'm finding I have to roll it all the way down sometimes. i have a Focusrite Solo and I'm using Reaper DAW with NeuralDSP Plugins and PAF pickups. Clean or high-gain, my noise floor seems out of control. Any advice?
@@ericsplittgerber5154
First of all, are you using new strings? Old, corroded strings tend to produce a lot of string noise.
Secondly, when you say "noise floor", do you mean the pickup's amplified self-noise or something else (like the string noise you mentioned)? The Focusrite has quiet preamps, so it shouldn't raise the noise floor to audible levels.
Thirdly, despite their name, not all humbuckers have equal hum-cancelling capabilities. High-output ceramic humbuckers with symmetrical coils tend to have the best signal-to-noise ratio, while vintage-output PAF humbuckers tend to have the lowest signal-to-noise ratios and limited hum-cancelling, due to the asymmetrical coils (one coil is typically wound a little hotter than the other, to minimize the frequencies that cancel each other out and to have a more balanced sound).
When adjusting the gain on your Focusrite, you should always aim for the strongest signal possible, without clipping. That's how you ensure you're getting the best signal-to-noise ratio possible. The more headroom you leave (by turning down the gain), the worse your SNR gets.
Then you turn down the plugin's input control to taste.
And lastly, amp sims emulate everything about an amp. Some amps are more noisy than others. And the higher the Volume control is set, the noisier it will get. Don't fall into the trap of always cranking the amp to full blast and then wonder why it's so noisy. The plugin's output level should be set by the Output control, not the amp's Volume control.
@@ErebosGR Hey thanks for the reply,
-strings are new, but I wasn't aware that old ones would have this much of an effect, so thank you for that.
-the pickups are specifically a Seymour Duncan JB , a burstbucker, and a Seymour Duncan Clean Machine humbucker, all lower output.
-I think I'm not using the term noise floor correctly here, but I mean that the pickups seem crazy hot and sensitive to everything I do on the guitar unless I roll the plugin input way down like -20 or more on higher gain amps. I don't get any hum from them when I'm not playing but they pick up the slightest movement on the strings and make a loud noise almost like someone is smacking their lips right in my ear. I'm careful to play with clean technique, and there's no open string sound, but theres an audible sound of me pressing the string into the frets, or of just lifting off that I never hear on a real amp.
I worry that I'm rolling the plugin input down too far to deal with this and I'm losing detail. It definitely takes a bit off the saturated lively edge of higher gain stuff.
-I'll mess with the output volume too, that could be part of the issue. I just always see people demoing these plugins and their input and output are always at a default noon and the sound seems so nice, lol. Makes me think I'm doing something wrong.
@@ericsplittgerber5154 The JB is not a PAF or a low-output pickup. It's actually among the hottest an AlNiCo 5 pickup can get. This could've been a factor, but you said that you don't have that issue when playing through a physical amp.
What you described is not a noise floor. A noise floor sounds like white noise when it's audible and looks like a floor on the spectrum analyzer.
It seems to me that you're just cranking the Volume of the amp too high, so it picks up and amplifies even the slightest of sounds (like pressing the string to the fret and letting go, like you said).
If, for your convenience, don't want to keep lowering the Input control on the amp sim for every preset, you could lower the gain on the Focusrite, but know that by doing so you are raising the noise floor (which may not yet be audible though, depending on how much amp gain you use).
e @ErebosGR Hey thanks! This was super educational, and thanks for clearing up the misconceptions. If the JB is hotter than I realized that's definitely part of the problem. I think NeuralDSP's Archtype Nolly also just runs pretty hot itself. I think It's best to just accept that I'll need to tweak things whenever I'm dialing in a new tone, and back things off for hotter pickups. There's a sweet spot where the string noise softens but I haven't lost nuance. Peace!
Great video. You should rename your video to "how to set up the input section for your VST". It might help people searching for this content. I spend along time looking for the answers you put in this video. You hit it right on the head.
As far as tone goes in my opinion, mics/mic placement and IR's seem to be the key. I've seen several videos where the presenter used "their IR" and there wasn't a discernible difference in side by side tone comparisons of demoed plugins. Any opinions on that?
Expected input gain , knowing what your input wants and what it will tolerate 🤘🏽
I came for the plugins, I stayed for the playing. Dude your taste is exquisite. Can you do a course, and if not, could you suggest resources to learn to play as you do
All valid points and once you get all of them in order but still struggle to achieve inspiring results, try this (given that you are trying to replicate something that's happening in the physical world and not trying to create something new):
1) Low/High pass filters are your friends. 50-120 Hz for High pass and 6-11 kHz for low pass usually give me best results but don't be afraid to experiment and trust your ear.
2) Reverb/Delay. Dry simulations sound unnatural since our brain is not conditioned to hear things that do not occur in nature so use reverb or delay. I myself usually have a block of somewhat subtle room reverb or plate reverb at the end of the chain (even if the chain has other reverbs or delays) to simulate the room. Again, try with big rooms and tiny ones, see what sounds best and trust your ear.
3) Compression. Add a block of compression after the cabinet simulation with very subtle settings. It should not be noticeable while playing and barely compress, this is done to simulate a loud cabinet in the room.
4) Panning. If you are using a stereo signal, pan it slightly to the left and right so it sounds a bit wider and not so directional since, in the real world, sound bounces off the walls and other objects. Again, trust your ear, but if you pan them out too much it will sound closer to a recording than an amp in the room. I sometimes like panning one side slightly more than the other to simulate an amp being a bit to the side, relative to my head.
When recording in the box it's worth the effort to consider a purpose built machine. I use a lot of fast RAM (128 GB). I have a high quality multi core CPU (Intel I7). If you are using samplers and virtual instruments solid state drives are where you want your sample libraries stored. All of what you said is 100% true. The small buffer size equals low latency. It does suck CPU but you can split different tasks to different cores. I have an old Tascam US-1641 24 bit USB interface. With that and my home built machine I can play with another guitarist and a bassist using 3 different instances of Helix native. We use Ableton live 10 (DAW), EZ drummer 2, and Kontakt (mostly keys) to round out the rest of the jam. Latency is never an issue. I've been chasing guitar computers since guitar rig 3 and I believe we are in the golden age of home recording. When I started I often wished someone would tell me what I was doing wrong and how to get better results. I spent a lot of time on forums and a lot of money on junk before I figured it out. I'm sure this video turned a light bulb 💡 on over a few heads.
Good advice 👍
Great advice John, I love your insight...all gained from experience..many thanks for sharing practical tips that helps us all with the amazing tech we have access to. With digital technology now available to us that tries very hard to emulate the analog sounds we like to hear, signal level, impedance and consequent latency are all very important indeed. Turn off anything that uses CPU time in a DAW, to make sure that as much as possible of the CPU clock processing is focused on what you are trying to do...this all applies to live or recorded signals.
This is helpful thankyou
Check out gig performer for a way of playing with amp sims as well as any vst plugins. It's made for live playing. No transport, no multiple tracks, no timeline. Rock solid. I use an rme interface, and get extreme low latency.
another tip for SOME plugins would be a highpass filter ... i sometimes get way better results with a 6 dB rolloff at around 75 Hz with my DI
Should always roll off high and low end you can't hear anyway. Improve your headroom in the mix
I guess the last section is due for an update (as you made us aware of). But good stuff in general, thanks!
As with one of your other videos I've just burned the guitars Ihad left.. Great videos.. Thank you sir..
What interface are you using now, and what routing do you use when using a Modeler? Do you pull your guitar input from your Audio Interface and plug it into your Modeler, or run everything into the interface and loop it back through the Modeler? Thanks for the tips.
hey john, this is a little off topic, but this youtuber made IR's out of the metallurgy cabs, hes giving them away so i tried them. I typically hate Helix cab sims and 3rd party IR's, but these are really phenomenal and easy to use because theres only 5 per cab. Gives a sneak peak at what 3.2 might be like
Really interesting video! I’m wondering if a nice transparent compressor with a buffer might be the perfect thing so help to get the input levels right?
No, as you'd get compression. Which I personally wouldn't want most of the time.
I've noticed my old Digitech Bad Monkey works very well for this method, as it's got both an "amp" & "mixer" output. Which I assume is both a Hi-Z & Low-Z option?
John Cordy for president!
I've seen a lot of YTbrs talk about input level and most of them say to get the signal to be to just below clipping. But I've also heard that you need to look at the level bar of the DAW track you're using and make sure it peaks at no more than 3/4 of the way. These are the optimal levels, as explained in Proper Recording Levels by Reapermania.
I'm baffled that ola sets gain to 0. Feels so weak when you play through that.
Your lead tone is amazing. (How are you getting that?)
Okay that intro was a flex
one thing i notice all by myself by accident is the fact that , in order to get your desired tone you have to spent countless hours trying to create your own sound, adjusting the mics height and the distance between the cab
so what worked for me and it did open "new horizons" was the fact that by accident, one day i open my DAW,added my tracks and i wanted to mount them as usual with my archetype petrucci plugin
all of a sudden i missclicked on one specific track and i did put a different plugin (i think it was the gojira one) and that was basicly it
now i do have such a thick and tight metal sound for my Drop B tuning :) i am happy, best mistake of my life tho
Is there a well established way of running the stand alone plug-in into the DAW. I’ve not found anything totally robust so far.
I have been using Bias FX for quite a while now and always hated how it killed all the hIgh end of my strats neck pickup sound. So it was the HI-Z feature wich i didnt know of until now... thank you dude!!!
i'm also a bias user so i feel your pain but i think i have some good sounding settings now.
What about using a DI box to run a guitar into a low impedance input? Is there any reason not to do this? I've actually found I like the tone of this better than simply running directly into the hi Z input.
I have the UMC 204 and D.I. Box Radial J48. My signal path is: DI>line input UMC 204 hd. I use the line. I must use the instrument line?
John there no difference for me with the Hi Z input of my old scarlet 2i2 gen1! (I tried the mini hall of fame which I read that has a nice buffer function)
What value -db you are opting to get in your input in DAW when recording? Read that somewhere around -10db it's nice so you have headroom for later adjustments in a song when mixing.
With my Fishman Fluence pickups and MOTU M4 I always go into the audio interface turned down. there’s a lot of signal there. Almost too much It’s a fine line. With the IBANEZ AZ I sometimes turn it up a bit. Also with the Ibanez BTB805MS active bass I turn up the signal a bit.
I plug my guitar in through the guitar jack of my Apollo twin x. From there I record into ableton live, will I be recording in HI-Z in this manner?
Also, if your PC has a powerful CPU, it might be in low clocks and causing pops/stutters. Set your PC power mode to High Performance and they'll all go away, as this stops the CPU downclocking.
^This. One windows search "sleep and power settings" and set it to high performance when doing audio. CPU usage on a daw can look low and you can get cracking and popping, not because the cpu is overwhelmed, but because it's trying to save power. This one change made huge difference to me.
How about a guitar with Active Pickups like an EMG 81. Should it be plugged into a Hi-Z input or Low Impedance input? (Btw, line level is equal to low level impedance? Thanks!
The latency was the main issue for me. I went through 4 different interfaces over the last few years and eventually got a rme babyface Pro interface and it, its usb drivers changed the game for me. Its able to take low buffer settings (low latency) but is very solid in that the play back is very stable.
1 issue I still bump into, is the rme will not gain down the instrument input, there's no pad or whatever so high output pickups are my new issue 😓
i went for a CPU upgrade xD the i5 14th gen runs about anything at 48kHz at 64 buffersize ... i loaded 23 instances of archetype rabea into ableton and it was "working" at 5ms roundtrip latency with the CPU at 50% i just stopped there cause it was getting rediculous lol
this should work on pretty much any interface if you use ASIO as your driver.
I had that same issue with the RME Babyface. I solved it by connecting the guitar into a direct box (Radial Pro48) which then connects to the interface in the line in input and uses the 48Hz phantom power. That input can be gained down to 0 on the TotalMix console. It worked great.
Hi, great video, question about impedence… do you apply the same concept of prefering High Impedence to the impedence settings on the Helix??? … I ask this because I have always thought that using a lower value on the impedence of the helix makes it sound better, less digital and more similar to an amp. I would realley appreciate your thoughts on this.
I think it'd be cool to choose based on taste rather than something that's technically "better".
Ola Englund commented on using Hi-Z input: input gain at zero - and you still get a good DI signal to be processed by plugin without any clipping
I know, that sounds so weird to me though. I know he gets better end results than most but it feels so weak playing with gain at 0.
Great advice, awesome playing in the beginning as usual!
But as you are a QC user, i have another question: as far as i know the QC doesn't have a Hi-Z input (or am i wrong here?). Do we have the same problems with missing high end too? For me personally a lot of the Factory captures sound a bit too dark, could this be because of a missing hi-z input? Could we maybe get even better sounds out of the QC with a buffer pedal in front?
I'm not a QC owner (yet), but the manual says that inputs 1 and 2 are Hi-Z when the input type is set to "Instrument". The impedance is adjustable from 10K to 10M. There are a few seconds about it from Neural DSP here: th-cam.com/video/KDlcPPClOEE/w-d-xo.html (starting at one minute, forty-eight seconds if my timestamp does not work for some reason)
@@jasonstewart759 ah ok, thx man!!
Great video, but it wouldn't be the same without that lovely intro jam. :D
I have an uphoria 202hd which has 2 inputs and there is a button that says "line/inst" and another one that says "pad" in each one of thouse inputs. Is there any chance that convers the input so to speak to hi z? Thanks for the video
So you want to go into the first input I think, and have it set to Instrument - that'll be the Hi Z I think
what audio interface were you using for the first test?
I just bought a Quantum ES4 and I'm having the same problem I always have with interfaces which makes me always get rid of them as follows:
You plug up your interface - you set the Mac audio to use the interface - you test the volume with some TH-cam videos or music - so far so good. You turn it up to a nice volume level. Then - you decide to plug your guitar into input 1 and play with some plugins or for the sake of this talk lets just say stand alone Tonex amp modeling plugin - not in Studio One or anything - just stand alone. It immediately sounds like crap - the reason is it's volume is so low. You check Universal Control - it's level coming across usb 1/2 is just as strong as the Mac system audio visually. So, why then is it so quiet? So, you crank the volume knob on the interface to compensate. You get it up to a good level for playing guitar. Now, then Mac system volume is way too loud. Why is this so hard? It should be easy - the levels of both are reading the same - but the actual output is way off.
There must be a way to boost the monitored plugin volume for Tonex, Neural Dsp, etc. Why is the volume they are outputting - although reading fine in the meters - so much physically more quiet than the Mac (or Windows) system volume coming across?
So - you take the opposite approach from above. You crank the volume of the interface to try and compensate for the quiet physical monitor volume of the plugins. Then - you separate the Mac system audio to Usb 3/4 and then take that volume down. So you've sort of come up with a compromise. Except that now - when you actually try and record - the volume you hear yourself record at when monitoring is louder than when you playback the recording. So irritating.
I have the same issue with every interface - so it's not unique to this - but why - why is this so difficult. You'd think plugging in your guitar and playing a plugin at a reasonable volume would be the easiest thing in the world. Every person in the world that tries this is going to do exactly what I did at first - get the system volume running first - thinking that sounds good - then try a plugin and be disappointed.
There must be a better way.
For those of us who dove in “early” in the DSP days and are BIAS FX or BIAS FX 2 users? I have a love/hate relationship with the product and could use some guidance, still.
I could listen to the 24 hours a day and not be bored.
hahaha I'm finally going to find a use for my Boss ML-2 pedal 😂
that's my own question, how can i get a real good tone with my plugins Ndsp
"The Guthrie is strong in this one." - Yoda
Are you using the Scarlet Line Level or Instrument level switch?
I have insane hum on my single coils at Instrument lvl input.
The hum is quite likely to do with being sat really close to your computer monitor?
Omg YES. That was it!. THank you!!!@@johnnathancordy
If you don't use higher input levels you don't use the full dynamic range of the A/D converter.
I have a couple of neural dsp plugins. It's the latency that gets on my nerves. I have a old line 6 toneport gx Interface through a laptop (AMD Ryzen 2.10gHz processor, 8GB RAM). I get all this artifact noise unless I go in the settings of the standalone program and then its latency city. The other thing is using the guitar volume to clean up the sound feels crap compared to my boss katana amp (which isn't a tube amp)
Yeh I've always found that Audio Interfaces integrated into technology like Helix/Fractal etc underperform compared even entry-ish level dedicated audio interfaces e.g. Behringer UMC
@@johnnathancordy My laptop probably isn't that well specced ( I don't know that much about it). If the interface does make a difference , could you (or anyone) recommend me a decent one that isn't too expensive? Oh, and love your videos by the way.
are you using ASIO as your driver?
@@ericlinfoot7419 Focusrite Scarlett Solo 3rd Gen is a great choice - and it’s very competitive on price😊
@@ericlinfoot7419 the MOTU M2 is cheaper than the Scarlett and has slightly better specifications with a way better metering display. I have the MOTU M4 which just has two more inputs and costs a bit more. I also use the Scarlett all the time and it is great, but if I was buying one again I would get the MOTU.
Don’t know if I’m hearing things but I have 2 laptops with the same Neural DSP and Guitar Rig plugins and the tone is way cleaner and crisper on the more expensive laptop . They basically have the same settings , same audio interface , but one sounds way better. Then again the worse sounding one is pretty damn old
This idea you're supposed to set the input gain on your audio interface to as high as you can without clipping has been discredited now - even though several interface manufacturers and amp modelling or plugin authors suggest this is what you should do - ignore them. For the most part you want the gain on your interface set to its lowest setting to get the right input into the plugin to match the same response as going into the real amp. Specifically though you want to look at the spec sheet for your interface for the max input level. If that's 12.2dbu then keep gain on min, otherwise adjust by the difference.
You should try guitar rig 6 pro
Does the buffer rule apply to 3rd Gen Focusrite Scarletts?
Yes, it applies to all audio interfaces using amp sims. Only interfaces with built in DSP amp are not affected. The focusrite does not have DSP
"Right, so..."
Ah, another good day :)
You know that you can buy STL plugins righ away, righ?
Also, only Tonehub and Controlhub have the subscription moderl. Tonality stuff does not have that.
Great advice. I prefer not using the Hi-Z settings on my mixer as like you mentioned, some of us don’t like a lot of treble.
Your advice, on the other hand, is really bad.
You should have a low-pass filter in post-processing regardless whether you like a lot of treble or not.
Having less upper harmonics at the start of your processing chain means that the amp sim and effects have less harmonics to amplify, thus producing a very bland, less musical sound with fundamental frequencies dominating your signal.
@@ErebosGR I’m not giving advice. I’m stating my preference. Where did you get that from?
@@tonepilot You spoke as if on behalf of "us who don't like a lot of treble".
@@ErebosGR You said I was giving advice. Never mind, you're trolling.
Fun fact: if I have output to my aiepods, I have latency while tracking
John!!! Turn your click off! :) Thanks for the vid sir, appreciate it.
Filter the top and bottom of your sound. Hi pass at 80Hz and Lo pass at 8kHz for starters. Adjust to taste. The top end of digital gear is what really makes it sound so different from an amp. There’s not much happening above 6kHz (give or take) coming from a guitar speaker cab. If I’m placing guitars in a mix I might Hi pass up too 200Hz. The main point is midrange is where the guitar lives and most of the useful (toneful) energy is there between 200Hz and 4kHz (again give or take).
if your into metal chernobyl studios has a great online course on this
I much prefer modellers over plugins just because this whole CPU load/latency thing. I have a decent laptop with i7-9750h processor, Steinberg UR22 mkii audio interface, latest drivers etc. and latency is still unacceptable and when I go lower I start to hear all these nasty artefacts like from overloaded CPU. I really hate laptops and Windows in general for this stuff. Apple computers are much better in terms of handling all these plugins. When i’m rehearsing with my band, my band mate’s MacBook is processing two guitars and bass in real time with automatisation and a bunch of other plugins and handles it like a champ.
I’ve used plugins before, but I’m in the privileged position to use tube amps, so that’s what I’ll go with.
any boss pedal will do that.
"all BOSS pedals include buffer circuits" - from Boss - so yep looks that way!
interesting I often have the opposite problem with input level in that some vsts react better if you back off as they are designed to want more headroom. When you push them too hard they get choked. It may be more of a thing for emulations of vintage gear combined with giving yourself room for gain staging. Related, the Helix blocks all behave like buffered pedals automatically. I have been working DAWless for a couple of years in order to focus on improving my playing instead of production. Cheers. Michael B
even bypassed and unpowered or does it have to be turned on?
@@benjaminvillarreal1499 for Boss pedals on or off.
@@benjaminvillarreal1499 if a boss pedal isn’t powered (either with a 9 volt battery or a power supply of some sort), it won’t transmit any signal because it’s not true bypass. But if the pedal has power, the buffer is in play whether or not the pedal is engaged.
You just gotta tweak on them. Good results for you won't necessarily be good for me because we all hear things differently at different frequencies.
WASAPI Shared is your friend. 48k/24bit is your friend. Almost like having computer puppies.
That is really horrible and even harmful advice. On Windows, pretty much every audio interface comes with drivers that support ASIO for lower latency. People will read your advice, use WASAPI and then complain that their latencies are bad, think their gear is faulty and send it back or even throw it away.
@@lik_doyaya you need to educate yourself a bit better before you go on the internet and write stupid things.