How to get Low Latency On ANY Audio Interface (ASIO) 

Playing guitar with big latency makes it almost impossible when the sound coming out is not what your playing. even a tenth of a second can realy throw you off. Im sure it would happen to drummers too, cause how can ya kep the beat when the beat is half a second off. Its weird. Thanks for making this. Ill be able to record on my computer again instead of using a lil pocket recorder that picks up every noise in the house.

One tiny clarification, It won't make a lower quality of audio by lower the buffer size unless the cpu gets overloaded. There isn't a difference in your voice at 5:00. You say "you can hear distortion and all that stuff" With maximum humility and respect for the rest of the great video, that isn't what you meant to say. It means the cpu will get overloaded and crackle with less load on it. It doesn't lower the quality of the audio though. The sample rate and bit rate control that, not the buffer size. Larger buffer means you can rock more vst's at once with more tracks and effects before encountering audio glitches. Once the buffer overloads the cpu will cause major audio crackle and drop outs sooner with a lower buffer. He just said it wrong because right after he says you could record audio directly in and have little latency. So he knows this is the exact time he would have it set to a super low buffer. You'd do this for recording vocals and turn it back up when you have like 3 dozen vst effects and synths going and you're mixing the beat down. I don't know crap about actual song production compared to him and am only clarifying. Nuff respect for your works!

Great explanation. Cleared up a lot of questions I had. Yes, trying to get the lowest latency might not always be the best choice. You nailed how the trade-off works. Thank you!

I swear Ave break it down better than most producers! I realized this when I just started messing with audio settings.

This really helped. I just upgraded from an old Korg interface that I never had any issues with, now I have the Kontact Audio 6 interface and the latency/glitching is really bad, but i will look into this tonight and see if I can use what I've seen here!

Thank you for your in-depth descriptions and explanations. You made it easier to understand, thank you.

I have a presonus 2/4 and an avid mbox. the First thing i noticed is the 2/4 handles latency better, but now i'm rethinking the mbox and i will try the techniques you've shown. Thanks for the knowledge

Thanks man, really well explained. I had a couple of synths that crackled, did what you suggested and work fine now!

Thank you so much. This was massive help! Keep up the good work man

Finally someone broke it all down correctly I been doing all this and known all about this still I see other people having issues about this subject basically if you play around with the settings and know what and how when to change the settings you will have no issues or problems like me I have no issues or problems I just know what to do what settings to play around with and adjust just like in this video I learned over the years so this is something newbies should watch most definitely and for anyone really to watch thanks much appreciated I don't suppose to be commenting I had to on this one this important lol stay blessed bro keep doing you 🙏🏾👍🏾🙏🏾

Enjoyed this video. Thanks for explaining this point. I will definitely like at this in my DAW more closely for the best results.👍

you're amazing man, thank you. fixed my tearing issue daaamn that was way simpler than other people were explaining. THANK YOU!

The most helpful video I've seen on the subject

Exactly what I needed since i switched to another DAW for my really powerfull pc instead of using logic on my old slow mac, the new daw didnt use ASIO as default. Very informative!

Thank you for the video, just setting up my AISO and this was extremely helpful.

thank you so much, it really took me this long to fix my latency issue. you're awesome

the whole page of settings and latency shifts or plug-in instability is just annoying where I've thrown out whole song ideas cause the track sounds broken or I've transposed a bunch of stuff and forgot what I wasa trying just keep making music anyway

Thanks for the video. Clarified what I understood, very informative 👍

Super explained! Simply great!

Sorry Ave, a little correction : 26.7 ms is obviously almost 1/40th of a second (not a little over a quarter second as you said in the introduction around 2:50). Cheers!!!!!

I can’t even move the buffer length bar, I’m on my brand new pc. It always happens when I get a new computer or something and nobody has a video explaining how to fix it

Sweet thanks! very helpful video i appreciate you for posting this.

Oh my god... That feels so good

congrats man, u'r the only one who realy resolve this s**** problem!! tks

Great video...thanks! Helped me out!

Great content, explained a lot for me

Hey,i want to know how to build channels to test the interface latency.

Thank you very much man this video worked for me 💯%

a quarter of a second is 250 ms

Great video sir- I am in my 50's just getting into Ableton live- and electronic music beginner. I am not able to change my buffer output size at all. Is it because I am only using my Pc driver? MME?DirectX driver? my output latency is 42.7ms and it's driving me crazy...

Always supporting ave 💪 gave it a thumbs up before the video started

You just made my day!

Wow..wow..thank you sir..really helped. Im on fl20 & maschine2k...

very clear ! thanks a lot

Would this work on mixcraft studio? I can not play back my audio without hearing loud scratching n popping while it plays. I add on neutron 3 and my stuff scratches and pops crazy

i did what you told and still got a 21 overall latency the in out sample rate wont go to 48000 if i turn it up to it, ableton freezes and my asio stop working is it my laptop that doenst have enough power or something? what can i do now?

Highly appreciated

thanks dude, what a champ!

I have an Avid fast track dou from ten years and I 9 milliseconds from the start each way. Is 9 good in n out?

I'm having a lot of trouble with latency on my medi controller, I played with the settings all day yesterday nothing seems to work I have a powerful computer rig but I'm using midi USB, would you recommend connecting through Midi cable onto my focus right and then connecting that to the PC?

I have a roland rubix 44. In the audio interface software i cant change my buffer size, so ableton still has a huge latency.. Someone knows how to fix this?

Nice stuff there thank's a lot

Thanks big dawg

Hi , how bout omec teleport what sample size you wud recommend? Tnx

I’m having a hard time understanding why a tenth of a second latency is tolerable. Like do you lower the latency when recording drums?

Peace bro could you do more mixing tutorials on Ableton?

thank you so much!! Got my latency fixed 2 min into the video lmao

so, could you begin a project and record audio first at a low buffer rate, then save the liveset, then increase the buffer rate and introduce plugins?

thnx heap man. great bro

I like a buffer of 48. But i have a new PC. I think it really comes down to what the PC can handle without tearing. Anything more than about 10ms latency is distracting to me.

Hey, thanks for addressing this subject!. I've got a noob question. So I've got a "Focusrite solo 2nd gen" interface right? I'm using Ableton on my brand new Lenovo Legion 740. Now, I haven't got studio monitors, so I basically use the interface to plug the mic. I used to work with Mac and never had the problem I'm having now, which is that I cannot hear a thing the moment I select ASIO as the controller type...why is that? I can see the signal comming through and all but I just can't hear it...does it have anything to do with not using monitors?. When I change the controller to MM/Direct X I do can hear but it's impossible to fix the latency problem...like, driving the buffer size to the minimum won't work...works wonders with MIDI but when recording vocals I then have to bring them forward in the timeline so they fit well...and well, obviously recording becomes a pain in the ass with latency busting my.... I really don't know what to do about this....thought with a brand new Windows computer perfectly meeting the specs wouldn't cause me such nightmare....I'm missing the easy "plug and play" Focusrite gave me with Mac....

What audio interface is best for live performance? Live gigging VST with midi keyboard controller... Best ASIO driver... Budget audio interface and yet very portable. I don't need good preamps, or how many input channels... Can someone tell me, share experience...

Latency party was very interesting

I need help! My ASIO Buffer Size is stuck on 512 Samples and it won't change! Please help

Can you do this same video but for FL

I think you want to click "mix in buffer switch" with an interface, even though I can't figure out what it does.

I lower the buffer size and I get TERRIBLE feedback. However, I higher the buffer size and it just keeps a load of latency. this sucks.

Ave, this was an excellent video clearing up an area of confusion for many since ASIO latency is a gray area for many. However, you made a comment that sparked something. I can't stand people who complain about their hardware not supporting their work. As a PC geek/nerd AND gamer with many decades under the belt (and I can't stand Apple/Mac) I get baffled (and a little irked) when someone complains about their machine not being able to handle the music workload. As a buddy of mine who was a pilot used to say, "You fly cheap you die cheap." Which I took and extended it to PC configurations. Buy cheap, you suffer. So, what I do as a PC builder/configurer for many clients and myself, is first sit down with the person and ask them to explain to me their work flow and what they want to do on a computer. From there, I am able to develop an idea of what hardware configurations will support them. And nowadays (thank you AMD!), processors are far more powerful and a lot less expensive than in previous times. I then tell them to give me a budget and warn them that I will be defaulting to the higher-end of their budget and not the lower end. At that point, I then ask them if they want to continue with me on this. If they're tightwads, they move on. If not, I'm able to show them what a difference a little more investment in their hardware can do for their workflow. I'll give you two examples. I have a sister who is an accountant and runs her own business. She was getting hosed by fake/greedy tech support people and buying pre-configured garbage "business grade" PCs. She called me up and I did with her what I explained above. We configured a laptop to be "gaming grade" and then I had it custom-built and sent to her. After I got it cleared/cleaned of garbage software and installed all her programs I told her to work with it and come back to me in a week. It didn't even take a day. She called me back gushing at how quickly she could work on everything! Trust me, as my accountant, she's customer #1 so that made my day! The second example is merely a comparison of machines example. My job has issued me a (Dell of course) "s"laptop specced Intel i5-6300U 2.4GHz processor with 8GB RAM. Sad indeed. I have another "Dell" laptop, that is 10 years old (Alienware 11inch of course) that is specced at Intel i7-2617M @1.5GHz process with 16GB of RAM that runs circles around my job-issued laptop. Why? Because I configured it and paid just a little bit more. Which brings me back to my point, if you want good hardware for your demanding software, don't go cheap. Duh. Too bad a lot of people skip over this point in their electronically based music endeavors. As always, thank you very much for sharing!

Thank you for info fam.

Tried what you just showed and I still have a big problem. I press a key on my synth and there is a delay to hear it from the computer. Plz help me fix this. Thanks

Recently bought an i7 core thinkpad purely for the purpose of producing and recording. Even with my buffer size set to the minimum 64 I get significant delay and latency

Please Read!! Urgent! I switched audio driver on ableton to asio and then i played this tutorial with sound going through headphones but all there was was this loud crackling in left ear. I tried changing the audio driver in ableton back to default and all audio i play is no static but its now really low quality, quiet and no bass. The only thing that is still normal sound quality through headphones is windows system sounds. everything plays back fine through my laptop speakers but not headphones devices. Please, If anyone can or might know how to fix this, Please reply!

Thank you!

What's a deal breaker is how unenjoyable it is to play with latency and it's why I still haven't invested in vsts. I'm looking to play an instrument, not tinker on my computer

hey man can i ask if the soundcard with condenser mic is applicable for recording? i try all type of shit and i dont know what shit going, i usually recording i hear my voice when i speak but when i play the recording in any DAW usually has no sounds can u help me dawgg?

I thought this was going to help me, but on MacOS there are only CoreAudio drivers. No matter what I do to the sample size or latency settings, there is a minimum of around 0.1 second delay between audio input and monitor. This is impossible to work with for live audio recording.

thanks which buffer size and sample rate should i use for fl studio with a focusrite scarlet 2i2 and sometimes recording on a good pc with an overclocked 3700x 32gb ram with 3600mhz and overclocked 3080 and 2 tb nvme m2 ssd?

1 am able to get 2ms with 48k sample rate with my laptop integrated sound card. That's pretty amazing. While my external sound interface can't achive this.

plz question! when i record am using 512 as latency on adobe audition , can i change it when i do mastering? its make a difference?

Thanks man

worth mentioning that even if the audio quality is higher at 96 or 192kHz you will not hear a difference. 44.1 kHz is already more than the human ear can resolve. higher sample rates come in handy if you slow down recordings or similar.

Wow - trying to imagine using my VSTs or even just monitoring through the sound card at 512 samples/26ms. That's VERY apparent and makes timing difficult. Everyone has their own sensitivity to this, but I'm surprised that you could perform with the latency set this high, unless you are just not monitoring through the DAW while you have it there. I can feel the difference between 64 and 128. Anything higher starts to get uncomfortable to me.

Thankyou for this explanation, I was trying for weeks to try resolve my issue and thanks to this video I've managed it My husband who knows nothing about this sort of thing was trying to butt in so I kicked him out , that's how bad it got . Might let him back now so you saved my marriage

super dope intro

Hiya. I'm new to all of this and I'm getting conflicting information and you do appear to understand this. I have a laptop (Acer A717 72G) gaming computer. The DAWs I will be working with are Cubase and Ableton Live (Lite.) Cubase is down right now due to a Windows issue with the e-licenser. I have two midi keyboards, miniLab MKII and Essential 49, both Arturia. I also have a Focusrite Scarlett Solo audio interface. When I was having trouble getting sound, someone recommended I use an ASIO driver so I installed ASIO4ALL. Then someone else told me never use an ASIO driver on a computer that doesn't have a soundcard. I am having issues with latency and 'someone' told me I am not able to resolve it because my processor isn't right. I have an Intel i7-8750H @2.20 GHz. I have no idea what all this means. Can I use an ASIO driver on my laptop? Would purchasing an external sound card make a difference? I know I'm asking a lot but if you can clear this up, I would be most appreciative. Thanks.

You said in any DAW to use ASIO. ..... yeah asio sucks on the razer blade. I cant watch a youtube vide and ableton at the same time. And i think because of it i got a bluescreen.. then when razer fixed it my audio stated craking but they dont with headphones. I now use mme/direct x

Are you using ableton live 10?

Im using presonus studio interface but my driver types only ever said core audio? Never seen ASIO before is that normal?

I just use WASAPI Exclusive mode, it feels like 0 ms.

Anything more than 5ms RTL is no bueno for vocalists, especially if you are stacking plug-ins and the vocalist is monitoring with the post processing.

It seems ridiculous that we still have to deal with MS and Apple not fixing this year after year.. in Windows the best way to solve this is to use Latency Monitor which will give you a list of the driver issues that cause audio issues...

nice!

My PC is brand new and really powerful. However when it set the buffer size to 512 I get A LOT of latency. And for some reason I can’t set it to 64 cause I get cracks and clicks. I am using asio4all and a focus rite Scarlett 2i2 Any suggestions?

Guide appreciated, only real comment is that one millisecond is actually a thousandth of a second, rather than a hundredth. Therefore... 26ms = 26/1000 S

hahaha i cracked up @0:34

MY ASIO driver doesn't work in my interface how do I fix it?

good desktop + pci-e card (with audio patching) + audio interface = low latency = better latency even after tracking a lot of tracks

26.7 ms are not a quarter of a second. 250 ms are a quarter of a second and very noticeable as an echo.

Increasing sample rate will lower latency. If you are not time-stretching your audio there's no need for anything higher than 48 kHz audio quality wise. You cannot hear the difference. Also, 26 ms is not a quarter of a second. 250 ms is.

4:00 Latenacy? Lol

Zero latency isn't possible. I hear a lot of content providers say this but it's not correct. Latency is the amount of time from trigger or creation of a sound at source until its heard from speakers or headphones. The human animal can detect a latency as small as 10ms. On average most non musician/audio engineers don't start detecting latency until it gets above 25-30. Professional audio engineers or musicians are 10-20. The trick with latency is not to try and eliminate it, but to get it to the point where it is so low that it's not detectible. This depends on whether you're dealing with Audio from outside the computer or audio generated within the computer like from a VST/Plugin synth of some kind. Latency is greater with audio from outside. Audio from a microphone, guitar or audio cable plugged device. Because the signal travels from the device, thru wire, into the interface, gets converted, to the DAW, processed and then sent back out to the speakers or headphones. Depending on how fast all of that happens, will determine the amount of latency. In audio, the use of audio buffers , which work with latency compensation algorithms, latency can be greatly reduced. Buffers take time to fill. So the trick with Audio is to make the buffer size big enough where the dynamic range of the audio can be handled without losing anything but not too big where you have a lag from the buffer taking time to fill. With VSTs, the audio driver or input/output conversion (I/O) part of the software is usually the problem. When you're recording Audio from outside, make the buffer small enough to reduce lag, but big enoug to not lose audio quality. With VSTs, it's the same thing, but you want to try and go as big as possible. Audio I usually use 64-128 VST/Plugins I use 128-512

Hey guys, a few thoughts. 1:56 "So I can select up to 192 kHz and that will improve the audio quality." NO, it will NOT. Well, in specific use cases it will, but as a general rule, anything that is beyond the human hearing range is literally irrelevant. There’s a FANTASTIC video about sample rates by Dan Worrall and I HIGHLY recommend everyone to watch it, but essentially: the Nyquist theorem explains why in order to cover the human hearing range we need sample rates of double the highest frequency that we can hear. 44,1 kHz is enough for that. This has nothing to do with the kind of compression such as mp3 where you may not be able to hear a difference between compressed and uncompressed: in this case there are measurable differences in how your brain reacts to the audio quality. That is NOT the case with audio extends beyond your hearing range. Your auditory sensors don’t register whether there’s anything there or not, end of the story. Higher sample rates are needed when playing back at slower speeds (works basically the same as FPS in video) or when using plugins that add distortion. 4:55 No, there is no difference at all to the sound of the audio being recorded with the microphone. Also, there is no distortion added. Ave makes it sound like you have to be worried about subtle audio quality loss with low buffer sizes. But really, you can only get that tearing he talks about. You obviously wanna avoid that. If you’re not gettin that, your’re good to go. 5:55 That’s a tricky one. Some plugins use multiple cores, some don’t. Fundamentally, each audio stream (e.g. a track or bus) uses one core. So it really depens on what you’re doing. Having more than 4 cores may not add much to your audio processing power. Anecdotal evidence suggests that raw single core speed is much more important in audio than the core count. What’s even MORE important though is your DPC latency and than can vary wildly from one Windows computer to the next because it depends on the hardware used and the drivers that are installed. There’s a REALLY helpful video by Richard Ames Music (search for "CPU performance vs. real time performance"). Finally... unfortunately this video didn’t really cover how to get low latency on any audio interface. I mean, everyone has their own idea of what "low latency" is and when we get into single digit territory, differences might be minute, but can still be huge if you think of them in percentages, and I think they matter. I’m trying to find out what the bottleneck in my system is so I can push the round trip latency below the 10ms mark, and I’m nowhere closer to that now after watching... :/ When you’re going to record MIDI and quantize anyway, higher latencies are perfectly fine, but for other tasks they’re not. Basically, I didn’t learn much apart from the fact that with Ave’s particular computer, he needs pretty large buffer sizes for stable playback...

At 2:50 you say 26ms is a quarter of a second. A millisecond is 1/1000 of a second.

26ms is not over a quarter second :) That would be 250ms!

2:48 250ms is a quarter of a second. not 25

The higher the latency, the more delay you'll have between key press and hearing the sound. People should consider using lower latency during the creative part of music making, and higher latency when mixing or EQing, etc.

25ms 1/4 second??? 1/40

If you computer and audio interface have USB type C inputs, connecting via USB C to USB C will lower latency. Just buy a cheap cable from amazon.

milliseconds means 1k ms =1s

26 ms is not .26 seconds.