As a professional audio engineer who mixes A LOT of recordings made by musicians at home, I really appreciate you mentioning the recording level of "-12 dBm." For whatever reason 95% of home recordings are recorded way too HOT (loud). In this age of modern digital recording, there's no need to record with the meters in the yellow, with an occasional red led, or "over." Save space for headroom, people! It will sound better in the end.
Can someone explain to me, an audio noob, why this is a desirable thing? I would think you would want to maximize your dynamic range, going as loud as you can without clipping. Why is leaving 12dB of headroom a good thing?
@@Fuffuloo I agree with you, but just shoot for 4-6dB of headroom when finishing the mix. Theoretically, the "4-6dB" is for the mastering engineer to work with. These days, there are "loudness standards" that streaming services and broadcasters use. It's no longer desirable to get ones mix as loud as possible. Both the industry, and even the public to a degree, are trying to put dynamic range back into recorded music. The "-12dBm" I mentioned earlier is for recording single tracks, like vocals,mguitars, bass, keys, etc, NOT ones finished mix. Let me explain that if every one of my tracks was recorded at "full scale," or just shy of clipping, then my master fader would probably end up well below unity gain. I'm too tired to type any more tonight. I hope that helps.
actually, there's a huge difference between the track input volume, for mixing, and the input gain into your guitar amp, which has got nothing to do with mixing levels. this is just confusing the two. the instrument recording should be at the optimum level for hitting the amp at virtual unity gain, not at the convenient mixing gain, which would basically be your amp out, and has got nothing to do with your d.i. signal. there's a lot of people offering bad advice out there, because they can't differentiate the two. even as pros. that's why you never listen to everybody just cause they're in the business. i personally deeply distrust most pros, in any field, for that exact reason. it's one thing to be doing something, and it's another altogether to have subtlety and understanding.
speaking of which i thought you're making a reverse mistake about buffer size (larger = more latency, less power requirement; smaller = less latency, more power requirement). I know it's nitpicky but just letting you know! Great advice as always, though.
Big ole correction on the buffer size. A smaller buffer size means you need a lot more processing power. When shrinking the buffer you are essentially decreasing the amount of time the CPU has to do it's work, but if a CPU is too weak or slow it won't keep up.... you will get crackles, snaps, and glitches. A larger buffer means you are giving the CPU more time to calculate or process everything, and is much more friendly to older or slower CPUs. ELI5: a larger buffer size is like giving a CPU paper and pencil to slowly go through everything and double check their answers open book test style. A smaller buffer size is like asking a CPU to answer a 60 question in less than a minute, BUT if they don't answer 1 question in 1 second correctly or in time the teacher slaps them.
Yep. The downside to a larger buffer is the increased latency though. So when recording you want to go for as small a buffer as your computer can handle. Most modern computers don't have an issue with a small buffer as long as you take a few basic preventative measures though. I make sure that all my other apps are closed. And then I freeze tracks I'm not using. And with Midi, I bounce it in place once I've picked a sound. I also turn off all plugins on my master bus until I really need them. I try not to use plugins with lookahead engaged. Etc.
Pay attention to the left channel switch boards 123 i usually go two or three. Noisegate your amp feedback! Amplitube isn't ment to be quiet or ampless but you can make your own presets like i've done and my presets are authentic and accurate!
@@AudioUniversity You are welcome. Thank you for the informations you have been providing us. It is of great help in music journey while setting up speakers for live meetings.
Lots of DAW's have some kind of low latency mode, like Logic Pro, Studio One... With that engaged there is no audible latency, even when recording with a lots of plugins on your channel.
I basically need something for practice at home and to have the best bang for the buck pedalboard and rig wise. This seems smarter than buying a line 6 helix for $1,000 just for at home practice.
I use Amplitube 5 for almost all my guitar sounds on my livestreams. It's tremendous, but you have to dig much deeper than this video.. Critical things include output compression, dual-amping etc... Also Reaper's metering is 'peak' metering by default... a -12dB peak input is too low, that's barely a quarter of the clipping level.. Play your absolute damned hardest on your highest output pickup and set that at -3dB otherwise you won't get a good response. Alternatively -12dB RMS would be a good 'average' volume but you need to set the Reaper metering appropriately in order to measure this correctly
@The Lamb The correct level for a very good DI is -12dbfs peak to -6dbfs peak in the Daw, maybe maximum -3dbfs peak, but that would be a very special case, or for a very short time. The trick is to adjust your levels, until you achieve this dynamic in your playing, and when you have a soft picking attack you are around -12dbfs peak.
Yes. You will get exactly what is supposed to be get: muddy digital sound and will spend rest of day in attempts to fix it with plugin... Add reverb and delay. That will fix everything :)
What's a good system latency (combined input + output) to aim for when setting the buffer size, to get a good feeling natural playing performance? 20ms, 10ms, or maybe less? At a 48 KHz sample rate, the 64 sample buffer you set (assuming the same for input/output) is 1.3 ms in/out, or 2.6 ms system latency. That seems rather low, but if the hardware allows for it, why not. But what's a good upper bound (buffer samples or latency) in case there's a need for performance or to mitigate loss samples / glitches?
It depends on your hardware and the DSP that is running. I would guess that the typical round trip latency varies from 3ms to 12ms. As for when you’ll notice it, that’s a controversial topic and I believe it depends on the individual.
@Josh Herald 14ms is a good latency that shouldn't affect your guitar playing at all, it's exactly the same latency you have when you play with a real amp standing in front of it 5 meters away. If you can't play your guitar properly with that latency in your Daw, you'll simply never be able to play live with real amps, and you'll have a rare disease, which will prevent you from being a musician.
Set your buffer as low as possible. Any latency is not a good thing; it'll compromise the feel. As a rule of thumb: As low as possible when recording. When mixing, make the buffer high, so you get all the system resources for plug-ins.
Headphones are a good option if you don't have studio monitors. It's best to use headphones or speakers that produce relatively even sound across the frequency spectrum!
How do people usually deal with latency? Is it something you just get used to? It seems like it would drive me crazy if I was trying to play hearing myself a few milliseconds behind my strum. Are there tricks to this or is muting the sound a good solution?
Buffer size is always the key to latency. Small buffer for recording, larger buffer for mixing/playback. In addition to those you mentioned, I also use Guitar Rig 4 but I recently picked up the Diezel Herbert from Plugin Alliance. Awesome sound
@@AudioUniversity I meant just from watching your video and assuming i lowered it, there would still be latency. just less. Curious if people just get used to it. Does lowering buffer size affect in any way the quality of the recording, or only the quality of the playback from your monitors?
@@optipwr40 buffer size does not effect the quality of your recording or sound unless your CPU is overloading. Keep in mind that if you're on Windows, you want to use the ASIO of your interface/sound card to minimize latency.
Once the latency is under about 8ms, you don't really notice it anymore. 8ms latency is the equivalent of being about 10ft away from your amp (speed of sound).
why is it that that the less the buffer size you have gives me less processing power? I thought it glitches because the cpu was running more! thanks. Also have you tried neural dsp's plug in? I know they are not budget friendly but they sound really awesome to my ears.
I have not used Neural DSP, but I’ve heard from several people that they sound great! I’m not sure how to properly articulate the power/latency issue. Just remember that when you ask your computer to process with a smaller buffer, you won’t be able to run as much processing (or as many plugins).
No, when the buffersize increases, there is more processing power, but that results in latency. Decreasing the buffer gives you less latency, but also less processing power.
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.
To get no latency, connect my headset to the interface, not to the computer. I also use the interface as output to the computer, meaning all my sound from the computer goes out through the guitar interface. I'm now happy and wonder why I have been so many years fighting the delay.
Nice video. I don't really have any plugins yet. Have you used Waves GTR3? It looks good, but I've heard some people say you basically get forced to upgrade.
Great video and very useful for totally ignorant people like me - in this realm. Few suggestions, slow down when using the components as we are trying to learn everything. For example in Amplitude (5 - free) I cannot add the pedals ---, nor get the dialog to set input to input 1. Minor... spellcheck 🙂
Hi dude! I use the free version of AmpliTube and some other free metal amp sims and impuls responses. Does anybody know how I could route midi into the AmpliTube plug-in? Thanks for your content!
I believe you would have to record the midi in a separate track, consolidate it, and send it to a different channel with amplitube. I could be wrong tho.
Okay so I’ve been trying to get a solid answer to this for a while. I use Mixwave Milkman Creamer plugin, guitar to interface and then I play through some 100$ speakers I bought at GC… I love the plugin, it feels like my sound at this point, but whenever I go to jam with people I’m forced to use my Line 6 Catalyst 60 that I’m not a fan of and I feel like my playing suffers because I don’t like the sound I’m getting. I’ve also tried using it as like a power cab with the cab sim shutoff on the plugin but it still sounds like shit. If I want the sound of that plugin but with better sound quality and at a higher volume should I get the Fender fr12 or something similar? I’m quite confused and don’t know which way to go any help would be greatly appreciated
The best guitar apps don't simulate old amps. Look for apps that do their own thing. Rhino is very good for dirty sounds and 20th Century does good cleans.
The best guitar apps don't simulate old amps. Look for apps that do their own thing. Rhino is very good for dirty sounds and 20th Anniversary does good cleans.
I just found your channel it seems to have a lot of great information, thank you. I was wondering if you could make a video on the best audio engineering online courses available? Im not talking about private schools or other online schools, I mean legit courses that are relatively cheap, but teach valuable knowledge. Is LinkedIn learning or coursera good options?
Thanks for your great video. But I have a problem. For example, if I insert the Mesa Boogie plugin into the track insert, I can still hear the clean signal from the guitar. I have the Steinberg Ur44c as an audio interface! What's wrong? I use Windows 11. Is the Asio driver working? Rock On 🤘
At 2:10 you said to adjust gain according to the meters in the DAW and not the clipping indicator on the interface. Is that true? I thought I'm supposed to adjust it until it starts to clip on the interface's meter (orange on the Focusrite Scarlett)? Can you elaborate on that a bit.
Hey man, Great video. I just have one question I can't find the answer to online. Why can't I route my output through Bluetooth headphones connected to my computer? Just trying to save a buck and use stuff I already have.
hi, I have this problem with my audio interface (focusrite scarlett solo) and my guitar that I haven't been able to solve for months, maybe you can help me. Every time when working with fl studio I have to record a guitar part I connect the audio interface to the PC and switch from fl studio's asio drivers to focusrite's to be able to use it, but the use of cpu in this way increases to almost maximum, by activating the monitor on the guitar track then the CPU usage becomes so high that the program and the sound stutter, making it impossible to record the guitar while leaving the other audio tracks active, how could I solve this problem in your opinion? Nice video anyway
Has anyone had this problem with Amplitube in Ableton? When using standalone Amplitube the effects work on my guitar, but when i add the plug in to my guitar trac in ableton, nothing happens. No input, output frequency is shown even though amplitube has popped up
MY solution the shitty sound. After a lot of trial and error. It's all about running signal hot enough ! This is my routine 👇 Use your soundcard first! Make sure u don't clip. Put a VU meter on your channel. Set the VU to a prefered headroom (-16 to -12 Db). Use gain knob on your channel and hit the strings on bass/guitar. Gain up until the th VU meter hits +/-0. THERE! Now u have a good signal. Load amplitube on the channel. Choose prefered amp. Load a 2nd VU after amplitube on a 3rd slot. Same settings as the first vu. And check u still hit +/-0 . You should have somewhere around the same value on your track meter as the swing on the first vu when hitting strings the hardest ( not by any means). IF not, boost signal on amplitubes OUTPUT SIGNAL, bottom right corner. I came up with this some time ago and it made my recordings cleaner. Less floor noise and more consistant. Please, comment or ask if necessary😊 👌🤘
Good video. I need help. I can listen back to my guitar (cakewalk daw, asio); but the fx plug ins like distortion or delay, dont change the signal! I listen to the guitar clean still. Also when I playback the recording I supposedly did, can see the waves but cant hear what I just played. Anyone got a fix for this? Want to practice with plug ins pedals on
gain staging at -12 peak is way too low for the amp. you need to gain stage for the amp, not for "mix friendly volumes". that's your amp out volume - for mix friendly. i've struggled a lot with gain staging into the amp, because of bad advice, and all virtual amp makers seem to agree that you need to set the d.i. gain as close as possible to 0 without clipping - playing the hardest you will play on that instrument. and that seems to sound the best, too.
I just cannot find a smooth distortion/overdrive, well ok the Boss Pocket GT had one sound but the rest were meh. Why are all distortions too fizzy or crunchy like annoyingly so?
Hmmmmm..... So, following the most basic rules of recording (like setting the buffer size) and using the Amplitube's default sounds would make your " DI Guitar Sound Good". Great tip 🤔
The main issue with VGAs is that they give nothing but a distorted effect. If you are looking for that amazing clean Fender sustain...you are not going to find it. No clean compressed Southern rock there. Only Noise !!!
Meh, putting a mic in front of an amp is easier to me but, you need a decent mic and amp. I never have a problem with bleed and if you do, throw the amp in a closet or isolation booth. Also, of course if you live in an apartment could be a problem although, I have recorded guitar at a fairly low level and it turned out fine. Never had good luck with those simulators. Always sound like ass.
It's a different approach but miking a real tube amp is certainly not easier to do for most people. The sounds in great amp sims like Amplitube 5 are perfectly usable for most people. I've done some A-B comparisons and had bandmembers and other musicians and even engineers listen to both if they could hear the difference. Literally no one could. It's all about perception. There is a difference in tone, but with some great EQ and other tone shaping plugins (Waves has some great ones, especially for guitar) it'll get you there as well. Just differently. And I've heard some awfully recorded real amps as well... 😏
So, I'm assuming that a "plug-in" is just new language for 'software', much like the term "app." And everyone wonders why this is so confusing... Stop @$%king renaming everything.
As a professional audio engineer who mixes A LOT of recordings made by musicians at home, I really appreciate you mentioning the recording level of "-12 dBm." For whatever reason 95% of home recordings are recorded way too HOT (loud). In this age of modern digital recording, there's no need to record with the meters in the yellow, with an occasional red led, or "over." Save space for headroom, people! It will sound better in the end.
Can someone explain to me, an audio noob, why this is a desirable thing? I would think you would want to maximize your dynamic range, going as loud as you can without clipping. Why is leaving 12dB of headroom a good thing?
@@Fuffuloo I agree with you, but just shoot for 4-6dB of headroom when finishing the mix. Theoretically, the "4-6dB" is for the mastering engineer to work with. These days, there are "loudness standards" that streaming services and broadcasters use. It's no longer desirable to get ones mix as loud as possible. Both the industry, and even the public to a degree, are trying to put dynamic range back into recorded music. The "-12dBm" I mentioned earlier is for recording single tracks, like vocals,mguitars, bass, keys, etc, NOT ones finished mix. Let me explain that if every one of my tracks was recorded at "full scale," or just shy of clipping, then my master fader would probably end up well below unity gain. I'm too tired to type any more tonight. I hope that helps.
Thank you Sir !
actually, there's a huge difference between the track input volume, for mixing, and the input gain into your guitar amp, which has got nothing to do with mixing levels. this is just confusing the two. the instrument recording should be at the optimum level for hitting the amp at virtual unity gain, not at the convenient mixing gain, which would basically be your amp out, and has got nothing to do with your d.i. signal. there's a lot of people offering bad advice out there, because they can't differentiate the two. even as pros. that's why you never listen to everybody just cause they're in the business. i personally deeply distrust most pros, in any field, for that exact reason. it's one thing to be doing something, and it's another altogether to have subtlety and understanding.
These plugins can I use them on midi controllers ?
the fastest growing pro audio channel in the west.
Thanks for supporting the channel, AI R!
speaking of which i thought you're making a reverse mistake about buffer size (larger = more latency, less power requirement; smaller = less latency, more power requirement). I know it's nitpicky but just letting you know! Great advice as always, though.
Big ole correction on the buffer size. A smaller buffer size means you need a lot more processing power. When shrinking the buffer you are essentially decreasing the amount of time the CPU has to do it's work, but if a CPU is too weak or slow it won't keep up.... you will get crackles, snaps, and glitches.
A larger buffer means you are giving the CPU more time to calculate or process everything, and is much more friendly to older or slower CPUs.
ELI5: a larger buffer size is like giving a CPU paper and pencil to slowly go through everything and double check their answers open book test style. A smaller buffer size is like asking a CPU to answer a 60 question in less than a minute, BUT if they don't answer 1 question in 1 second correctly or in time the teacher slaps them.
Yep. The downside to a larger buffer is the increased latency though. So when recording you want to go for as small a buffer as your computer can handle.
Most modern computers don't have an issue with a small buffer as long as you take a few basic preventative measures though. I make sure that all my other apps are closed. And then I freeze tracks I'm not using. And with Midi, I bounce it in place once I've picked a sound. I also turn off all plugins on my master bus until I really need them. I try not to use plugins with lookahead engaged. Etc.
Interface and settings tutorials would be greatly appreciated and likely be very popular..
Pay attention to the left channel switch boards 123 i usually go two or three. Noisegate your amp feedback! Amplitube isn't ment to be quiet or ampless but you can make your own presets like i've done and my presets are authentic and accurate!
It would be nice to do a video talking about creating presets for live performance (levels, comp, eq...). Thanks
Thanks, JuLee!
That would be amazing
yeah with the title I was expecting a video on some settings in the software. This is more of a how to get started guide
That's what i was expecting lol...
We are using Amplitude with iPhone and iRig 2. It sounds amazing.
You can also play in live performances, using above rig.
Thanks for sharing, BINU SAM!
@@AudioUniversity You are welcome.
Thank you for the informations you have been providing us.
It is of great help in music journey while setting up speakers for live meetings.
Waves GTR 29.99 is awesome, all you need it has loads of effects, I built great presets in seconds.
Lots of DAW's have some kind of low latency mode, like Logic Pro, Studio One... With that engaged there is no audible latency, even when recording with a lots of plugins on your channel.
*This may be just what I need to conserve on space in an already small and crowded studio🤔🇯🇲*
Space! That’s another benefit! Thanks, Jerri.
I basically need something for practice at home and to have the best bang for the buck pedalboard and rig wise. This seems smarter than buying a line 6 helix for $1,000 just for at home practice.
I use Amplitube 5 for almost all my guitar sounds on my livestreams. It's tremendous, but you have to dig much deeper than this video.. Critical things include output compression, dual-amping etc... Also Reaper's metering is 'peak' metering by default... a -12dB peak input is too low, that's barely a quarter of the clipping level.. Play your absolute damned hardest on your highest output pickup and set that at -3dB otherwise you won't get a good response. Alternatively -12dB RMS would be a good 'average' volume but you need to set the Reaper metering appropriately in order to measure this correctly
Thanks for sharing these tips, The Lamb! Much appreciated.
@The Lamb
The correct level for a very good DI is -12dbfs peak to -6dbfs peak in the Daw, maybe maximum -3dbfs peak, but that would be a very special case, or for a very short time.
The trick is to adjust your levels, until you achieve this dynamic in your playing, and when you have a soft picking attack you are around -12dbfs peak.
You made the video we talked about 👍🏼
Absolutely brilliant and my favourite new TH-cam channel 😀
Thanks, Andrew!
look up SCUFFHAM SGEAR -> add an overdrive pedal in front of the interface -> ENJOY!
Yes. You will get exactly what is supposed to be get:
muddy digital sound and will spend rest of day in attempts to fix it with plugin...
Add reverb and delay. That will fix everything :)
What's a good system latency (combined input + output) to aim for when setting the buffer size, to get a good feeling natural playing performance? 20ms, 10ms, or maybe less? At a 48 KHz sample rate, the 64 sample buffer you set (assuming the same for input/output) is 1.3 ms in/out, or 2.6 ms system latency. That seems rather low, but if the hardware allows for it, why not. But what's a good upper bound (buffer samples or latency) in case there's a need for performance or to mitigate loss samples / glitches?
It depends on your hardware and the DSP that is running. I would guess that the typical round trip latency varies from 3ms to 12ms.
As for when you’ll notice it, that’s a controversial topic and I believe it depends on the individual.
@Josh Herald
14ms is a good latency that shouldn't affect your guitar playing at all, it's exactly the same latency you have when you play with a real amp standing in front of it 5 meters away.
If you can't play your guitar properly with that latency in your Daw, you'll simply never be able to play live with real amps, and you'll have a rare disease, which will prevent you from being a musician.
Set your buffer as low as possible. Any latency is not a good thing; it'll compromise the feel. As a rule of thumb: As low as possible when recording. When mixing, make the buffer high, so you get all the system resources for plug-ins.
I'm so happy I still have rocksmith usb cord because this amazing stuff
What pickup position should your guitar be set to to have the most flexibility to experiment with different pedals and amps in the daw???
Amplitude also has a android version you can use if you have the irig
I have the iRig Pro Duo I cannot figure out how to get amplitube for Android it's like it doesn't exist anymore
@@steveclark9934 I think it's mutated into tonebridge
Great explanation. Concise and going over all the important stuff. Thanks!
Thanks for watching, Miguel!
Running ezdrummer2 with amplitube5 and bias fx 2 is the way to go for big full sound!
actually there is one nembrini amp that really is a powerhorse I recommend getting that.
Do you need studio monitors to replace the laptop speakers? Or a headphones are good enough?
Headphones are a good option if you don't have studio monitors. It's best to use headphones or speakers that produce relatively even sound across the frequency spectrum!
@@AudioUniversity do you have any headphones recommendation foe overdrive and high gain guitar playing?
I’ve had good experiences with Audio-Technica, Beyerdynamic, and Sennheiser. There are many headphones that I haven’t tried which are also great.
So that is where the latency was hiding. Thanks!
Thanks man. You're helping me feel comfortable to getting into this all.
Glad to help, Bob!
this was the most straightforward, easy to understand tutorial. Thank you a bunch.
How do people usually deal with latency? Is it something you just get used to? It seems like it would drive me crazy if I was trying to play hearing myself a few milliseconds behind my strum. Are there tricks to this or is muting the sound a good solution?
Did you lower your buffer size?
Buffer size is always the key to latency. Small buffer for recording, larger buffer for mixing/playback. In addition to those you mentioned, I also use Guitar Rig 4 but I recently picked up the Diezel Herbert from Plugin Alliance. Awesome sound
@@AudioUniversity I meant just from watching your video and assuming i lowered it, there would still be latency. just less. Curious if people just get used to it. Does lowering buffer size affect in any way the quality of the recording, or only the quality of the playback from your monitors?
@@optipwr40 buffer size does not effect the quality of your recording or sound unless your CPU is overloading.
Keep in mind that if you're on Windows, you want to use the ASIO of your interface/sound card to minimize latency.
Once the latency is under about 8ms, you don't really notice it anymore. 8ms latency is the equivalent of being about 10ft away from your amp (speed of sound).
Is there any benefit to using a daw vs just the standalone amplitube app for desktop
probably more options in terms of editing your sound
Thank you for your upload it's easy to understand for dummies like me
Glad to help, Cory! Thanks for watching.
why is it that that the less the buffer size you have gives me less processing power? I thought it glitches because the cpu was running more! thanks. Also have you tried neural dsp's plug in? I know they are not budget friendly but they sound really awesome to my ears.
I have not used Neural DSP, but I’ve heard from several people that they sound great!
I’m not sure how to properly articulate the power/latency issue. Just remember that when you ask your computer to process with a smaller buffer, you won’t be able to run as much processing (or as many plugins).
No, when the buffersize increases, there is more processing power, but that results in latency. Decreasing the buffer gives you less latency, but also less processing power.
man, amplitube is really pretty
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.
To get no latency, connect my headset to the interface, not to the computer. I also use the interface as output to the computer, meaning all my sound from the computer goes out through the guitar interface. I'm now happy and wonder why I have been so many years fighting the delay.
Amazing video tutorial. I really appreciate it so much. Thank you man and keep up the awesome work.
Nice video. I don't really have any plugins yet. Have you used Waves GTR3? It looks good, but I've heard some people say you basically get forced to upgrade.
Yes. I’ve used it a lot. I’ve really enjoyed it.
Love Reaper.
Great video and very useful for totally ignorant people like me - in this realm. Few suggestions, slow down when using the components as we are trying to learn everything. For example in Amplitude (5 - free) I cannot add the pedals ---, nor get the dialog to set input to input 1. Minor... spellcheck 🙂
Thanks for the tip! I’ll keep that in mind for future videos.
STRAIGHT FORWARD NO BS.. THNX SUBSCRIBED
Thanks!
Thoughts on the amp sims that come with garageband ? Any video on how to make the most of out it since it’s free it’s likely to be often overlooked
I haven’t used them personally, but I’ve heard some good sounds made with them.
Great video thank you! But what about volume on the guitar? Always at 10?
Hi dude!
I use the free version of AmpliTube and some other free metal amp sims and impuls responses.
Does anybody know how I could route midi into the AmpliTube plug-in?
Thanks for your content!
I believe you would have to record the midi in a separate track, consolidate it, and send it to a different channel with amplitube. I could be wrong tho.
Okay so I’ve been trying to get a solid answer to this for a while. I use Mixwave Milkman Creamer plugin, guitar to interface and then I play through some 100$ speakers I bought at GC… I love the plugin, it feels like my sound at this point, but whenever I go to jam with people I’m forced to use my Line 6 Catalyst 60 that I’m not a fan of and I feel like my playing suffers because I don’t like the sound I’m getting. I’ve also tried using it as like a power cab with the cab sim shutoff on the plugin but it still sounds like shit. If I want the sound of that plugin but with better sound quality and at a higher volume should I get the Fender fr12 or something similar? I’m quite confused and don’t know which way to go any help would be greatly appreciated
is your metering showing you input or output level, pre amplitude or post
you don't need to adjust the input level, keep it at zero
At least in Amplitube 4 I could never get satisfying sound but maybe 5 makes the difference.
Use D.I. Will solve the problem. Don't use default inst inputs of interface
The best guitar apps don't simulate old amps. Look for apps that do their own thing. Rhino is very good for dirty sounds and 20th Century does good cleans.
The best guitar apps don't simulate old amps. Look for apps that do their own thing. Rhino is very good for dirty sounds and 20th Anniversary does good cleans.
These plugins can I use them on midi controllers ?
You would need to control an instrument plugin with MIDI and then run that audio signal through these plugins.
@@AudioUniversity right so. I get my midi controller connect it to the plug-in then connect those to the daw?
These plugins only work for audio signals. Check out this video: th-cam.com/video/4revAw3lT4g/w-d-xo.html
@@AudioUniversity ah so they modify already produced sound thanks I’ll watch that video I appreciate this
I just found your channel it seems to have a lot of great information, thank you. I was wondering if you could make a video on the best audio engineering online courses available? Im not talking about private schools or other online schools, I mean legit courses that are relatively cheap, but teach valuable knowledge. Is LinkedIn learning or coursera good options?
Great Video
I use amplitube 5
I can clearly hear the strings striking sound like “huh huh “ when I play the lead part
Any solution for this please ?
Ik im late but have you tried using a transient shaper?
Thanks for your great video. But I have a problem. For example, if I insert the Mesa Boogie plugin into the track insert, I can still hear the clean signal from the guitar. I have the Steinberg Ur44c as an audio interface! What's wrong? I use Windows 11. Is the Asio driver working? Rock On 🤘
At 2:10 you said to adjust gain according to the meters in the DAW and not the clipping indicator on the interface. Is that true? I thought I'm supposed to adjust it until it starts to clip on the interface's meter (orange on the Focusrite Scarlett)? Can you elaborate on that a bit.
I love music.
Hey man, Great video. I just have one question I can't find the answer to online. Why can't I route my output through Bluetooth headphones connected to my computer? Just trying to save a buck and use stuff I already have.
hi, I have this problem with my audio interface (focusrite scarlett solo) and my guitar that I haven't been able to solve for months, maybe you can help me. Every time when working with fl studio I have to record a guitar part I connect the audio interface to the PC and switch from fl studio's asio drivers to focusrite's to be able to use it, but the use of cpu in this way increases to almost maximum, by activating the monitor on the guitar track then the CPU usage becomes so high that the program and the sound stutter, making it impossible to record the guitar while leaving the other audio tracks active, how could I solve this problem in your opinion? Nice video anyway
Thanks for your efforts
Thanks for watching!
Has anyone had this problem with Amplitube in Ableton? When using standalone Amplitube the effects work on my guitar, but when i add the plug in to my guitar trac in ableton, nothing happens. No input, output frequency is shown even though amplitube has popped up
MY solution the shitty sound. After a lot of trial and error. It's all about running signal hot enough ! This is my routine 👇
Use your soundcard first!
Make sure u don't clip.
Put a VU meter on your channel. Set the VU to a prefered headroom (-16 to -12 Db).
Use gain knob on your channel and hit the strings on bass/guitar. Gain up until the th VU meter hits +/-0.
THERE! Now u have a good signal.
Load amplitube on the channel. Choose prefered amp. Load a 2nd VU after amplitube on a 3rd slot.
Same settings as the first vu. And check u still hit +/-0 .
You should have somewhere around the same value on your track meter as the swing on the first vu when hitting strings the hardest ( not by any means). IF not, boost signal on amplitubes OUTPUT SIGNAL, bottom right corner.
I came up with this some time ago and it made my recordings cleaner. Less floor noise and more consistant. Please, comment or ask if necessary😊
👌🤘
Do these plug ins work with pro tools?
Yes.
Good video. I need help. I can listen back to my guitar (cakewalk daw, asio); but the fx plug ins like distortion or delay, dont change the signal! I listen to the guitar clean still.
Also when I playback the recording I supposedly did, can see the waves but cant hear what I just played.
Anyone got a fix for this? Want to practice with plug ins pedals on
Which daw do you recommend?
What's the best sample rate to use?
gain staging at -12 peak is way too low for the amp. you need to gain stage for the amp, not for "mix friendly volumes". that's your amp out volume - for mix friendly. i've struggled a lot with gain staging into the amp, because of bad advice, and all virtual amp makers seem to agree that you need to set the d.i. gain as close as possible to 0 without clipping - playing the hardest you will play on that instrument. and that seems to sound the best, too.
I just cannot find a smooth distortion/overdrive, well ok the Boss Pocket GT had one sound but the rest were meh. Why are all distortions too fizzy or crunchy like annoyingly so?
Reaper 🙌🏼
Hmmmmm..... So, following the most basic rules of recording (like setting the buffer size) and using the Amplitube's default sounds would make your " DI Guitar Sound Good". Great tip 🤔
Very interesting. Thanks!
can it be used with virtual guitars?
Good videos still excist.
Thank you so much and subscribed.
Great job
Thank you
I assume this would work with a virtual guitar aswell? “Session guitar”
Wow. Well done. Subscribed
The main issue with VGAs is that they give nothing but a distorted effect. If you are looking for that amazing clean Fender sustain...you are not going to find it.
No clean compressed Southern rock there. Only Noise !!!
I've tried so many of these VST plugins and they always sounded muddy and horrible, I gave up.
I need the settings to sound like Van Halen.
What is DI? I thought it’s an audio interface ?
Check out this video, Michal: th-cam.com/video/_xybjiuD9K0/w-d-xo.html
@@AudioUniversity awesome! Thank you!
Amplitude is garbage. All they want to do is sell you more. Super annoying
It’s very distracting when things like a graphic with “guitar” spelled wrong in large type just 52 seconds in.
Should be called " beginner's guide to di guitar " or something
Neural Amp Modeler. The end.
But why does it sound so muddy ?
Amp sims always sound worse than no amp sim - i just dont get it
Reaper, yaay.
Meh, putting a mic in front of an amp is easier to me but, you need a decent mic and amp. I never have a problem with bleed and if you do, throw the amp in a closet or isolation booth. Also, of course if you live in an apartment could be a problem although, I have recorded guitar at a fairly low level and it turned out fine. Never had good luck with those simulators. Always sound like ass.
It's a different approach but miking a real tube amp is certainly not easier to do for most people. The sounds in great amp sims like Amplitube 5 are perfectly usable for most people. I've done some A-B comparisons and had bandmembers and other musicians and even engineers listen to both if they could hear the difference. Literally no one could.
It's all about perception. There is a difference in tone, but with some great EQ and other tone shaping plugins (Waves has some great ones, especially for guitar) it'll get you there as well. Just differently.
And I've heard some awfully recorded real amps as well... 😏
-12 db fullscale
So, I'm assuming that a "plug-in" is just new language for 'software', much like the term "app." And everyone wonders why this is so confusing... Stop @$%king renaming everything.
what about Helix Native?
I haven’t tried it, but I’ve heard good things. Is that what you use, Carlos?
@@AudioUniversity Yes! I really love its simplicity and versatility. Great emulations too!
Does Amplitube feature dedicated presets of sounds, which can be modified if the user chooses to do so?