How to RECORD with EFFECTS and WITHOUT LATENCY

How to RECORD with EFFECTS and WITHOUT LATENCY
If you don't have a DSP type Sound Interface or a digital mixer,
learn how to record and monitor your signal with no LATENCY and with the use of EFFECTS.
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[@jonowirion]

Hi chris great info! Until now i have been bouncing full tracks down to record vocals on top with desired effects etc then decided to search for a faster less hassle way to to it! The only part of this video i wasnt sure about is how to route the cue mixes but ive not tried it yet. Ill go and give that a shot now.

[@Bring_MeSunshine]

I'm going to have to try this. My interface's software mixer is Audient iD mixer which provides ultra low latency monitoring and 2 independent cue mixes, but no internal DSP plugin facility, and I know from experience that some singers want a bit of verb to get the mood right. In the old days (ahem!) I used Cubase VST 32 v5 and could record a clean signal, while still using an insert effect for the vibe, but despite improvements in CPU memory and interface drivers, the demands of modern DAWs on the system's resources has negated that facility somewhat. Excellent, very useful video, Chris, and judging fro the comments below you've also helped a lot of other Cubase users. Nice one

[@johnhowarthmusic8706]

Thanks, Chris.. As always, your video is straight-forward and simple. Mainly, the fact that I can relate to your logic, is a BIG plus, for me. I have Cubase Artist 9.5 and Steinberg techies, forums, manuals all seem to be of little use, or confusing - in a different language....lol (To me). I am about to return a UR44, which was going to replace my UR22. Two advantages being DSP and extra inputs. (I really only need three;: 2 stereo INs for keyboards and one for mic, which is a 'workflow/laziness' issue. (-: My UR22 has a blend/mix knob for blending vocal/DAW music tracks but no Direct monitoring (DSP). The UR44 has DSP but no simple way of balancing/blending my recording vocal with my DAW music track level. Your Lynx has a software mixer. The UR44 has, too but it's only meant to work with other DAWs!! Pretty inconvenient, if not dumb, IMHO. So, it cannot be used!! Steinberg techies tell me I need to either send my DAW tracks into a Group track separate from my vocal, or use Q-link for my music tracks to lower the levels against the vocal!.. Then increase the H/phone level. I hope this helps others and, maybe, informs you? I saw your "competition video" but (unfortunately, for me) you didn't cover these issues I'm told the answer is NOT to buy a 4 channel mixer for my UR22 (which would solve these issues?), because the preamp would be inferior to the UR22 = signal noise?

[@mixdownonline]

Hey John, I have the UR-RT4 that comes with the DSP Mix as well and with CUBASE there's a specific way to use the DSP system for monitoring...I was planning on doing a tutorial video on this. I'll bring this one on my priority list, maybe it will help you out

[@robertgaudreau2520]

Interesting, and tricky, if I’m right my understanding is; from the interface device, the input signal goes directly to the output of the device without having to pass through the DAW. Then adding on low latency within the DAW at the same time another output where the signal gets through the activated DAW monitoring. Great video Chris!

[@mixdownonline]

That's pretty much it :-)

[@rcan1958coxnet]

Very nicely explained. Now I will be solving my problems using direct monitoring. Also had no idea one could still put some fx on it for the singer when direct monitoring. I think the singer will very much more comfortable with a touch of fx on during the session.

[@Scotlanz]

Unbelievable! Exactly what I've been trying to do for the last two days! I finally figured out latency free monitoring in Focusrite MixControl but it's complicated. Hopefully when I watch this video later it'll give me a better option. Thanks in advance man! Edit - Well I just finished watching and it doesn't really do what I was hoping for. Or at least it does exactly the same as the focusrite. My usual way of getting rid of latency is to use the Delay Constrain feature which is great but you lose a lot of your plugin fx. I was hoping this was a workaround for this....which is sort of what it is, but no different really from using the DSP in MixControl. A good video nonetheless for anyone who doesn't have a DSP interface. Cheers et merci a Irlande.

[@mixdownonline]

Yes, if you have a DSP system on your interface, that,s the way to go. My trick is only if you don't have a DSP interface or a digital mixer. Thanks for watching :-)

[@landonjett]

hey Eric, I am sitting here looking to fix the same issue. trying to track vocals with no latency through my Scarlett 18i 20 2nd gen. any tips you can send my way? cheers.

[@matthias100]

Thank you man you are only one in youtube who teach in a very clarifying way

[@mixdownonline]

You're more than welcome, thanks for your comment!

[@92soldier]

brother you just solved theee most annoying thing ive ever experienced in FL studio. I literally just did what you did in FL and Ive been doing live shows with live effects with NO latency. Sometimes it shows 1ms, but it goes back to 0.

[@vainsacrosanct2014]

Thanks for the buffer tip for mixing. It was a helpfull tutorial

[@mixdownonline]

You're welcome!

[@RiteshGandhi]

Hi Chris..thanks a lot for all your extremely useful videos. Needed a piece of advice.i had an RME UCX which got burnt due to some power outage and I need to get another audio interface without any DSP effects as I don't need them and audio interfaces without DSP would be cheaper. Could you please suggest which audio interface should I go for . I need 8 in and 8 outs and run a professional studio and my budget is around $700-800. Thanks in advance

[@NeoONELove]

Brilliant! However I'm wondering for a plugin like autotune, wouldn't the effect be delayed by the latency. I understand it doesn't matter that much with reverb but would other effects be delayed...other more time sensitive effects?

[@MikkjalHvannastein]

Always great videos man, thanks. I have a question. Can i do this the same way with like Halion? . My buffer size is at 2048, and when playing it gets really annoying. Thanks Chris :)

[@mixdownonline]

You're welcome! To answer your question, it's not the same case with MIDI... at 2048 buffer size, that will be very hard to play your VSTi, bring the buffer size down and if that is too hard on your system, you can Freeze other VSTi while recording what you need with Halion.

[@phillwildman]

Excellent explanation. Thanks

[@mixdownonline]

You're welcome!

Nice man!! Thanks ❤️

[@mixdownonline]

You're welcome

[@MikkjalHvannastein]

DAMN 😱😱😱. Thats friggin awesome. Thanks alot Chris 👍🏻👍🏻🎸

[@mixdownonline]

You're welcome!

[@benroxxx]

How should I go about it if I am recording on Logic on my 2017 MacBook Pro with a Blue Yeti Mic? Having the effects on when I record renders monitering it useless because when I play back vocals, they are off beat.

[@krns1695]

I am using Cubase 5, and I was wondering if theres a way to add Fx from the DIY, instead of the interface, for example : record with autotune/reverb already on, so I can hear them in the headphones while recording.

[@GaityMan]

dude!!! 2020!!! cubase 5!!!! hahaha

[@drummerguyron]

I use Focusrite Mix Control for a liquid saffire 56. I only use output 1 and 2, for my reference monitors. So I'm assuming I just use the cue mix routed to 1 and 2?

[@mixdownonline]

I never worked with the Focusrite Mix Control but looks similar to what I have with my Lynx... You don't have to use a Cue Mix to do this technique. You can use the 1-2 Stereo Output and blend the Vocal Reverb Track with the dry signal from the MIX CONTROL...or just create a second Stereo Output in Cubase (3-4) and Route the output of the Vocal Reverb track to that Output. And like I did, control the reverb level from your MIX CONTROL to your monitors. You're just gonna have to learn how the Focusrite Mix Control works though

[@Dyonysus777]

Nice video and the others as well 😊. Intro: Let me say I have a Yamaha DGX 670 with a built in audio interface so no external audio interface, which however I am considering to purchase (more at "NEXT Subject") and possibly with DSP if you recommend it to me? I am not sure I got the conclusion: So you say your FX signal directly from the audio interface is the only signal we hear/monitor, but what if you want to hear only the FX but record the signal without FX effects: Do you then only record track 1 and 2 without recording the FX (well perhaps both simultaneously and then use the 1+2 track to add Cubase effects/inserts later)? SO when you record 1 and 2, then do you mute it while recording (but maintain the gain turned up), in order to only hear the FX, when recording vocals, or do you listen to both with as small a buffer size as possible meaning as low latency as possible. As far as I understand it, then when you record the 1 and 2 in CUbase then you hear it in the monitors but I thought by muting it you can record these channels and their tracks, without the FX effects. SO sing along with say REverb but record the the singing simulataneously without the effects. I wonder how you can get the effects without passing Cubase when you don't have a DSP on the Lynx audio interface?? SO that is thanks to LYnx? Seems like the same principle as DSP to me since you are monitoring the Audio interface directly, OR??? Normally when monitoring the signal with DSP directly from an interface like Steinberg UR44/Ur22, then you can only create a track in CUbase but it is not controlled from CUbase. BUT in this case you actually added the effect in the FX track. Hmmm I thought this would be impossible without DSP (in which case you add the effects at the audio interface using some software I guess since it is not an analog mixer). IS THIS method you use, what is called Native Low Latency REcording? Maybe I just don't understand really the essence of an FX track..hmmm. Never really used it. I just used inserts on the track itself. ALso I typically recorded with the effects added already on the piano (including on the Vocals), but not in exaggeration in order to later add more if needed (like 18 wet value of say medium hall). Can one do that with another brand of Audio interface or only with Lynx? What about Tascam Audio interface US-2x2HR USB 2.0 24bit / 192kHz (hesitating with that since its USB 2.0 so that would introduce latency I think). Or Scarlett FOcusrite something or any other recommendation? OR CAN I MAKE SOME WORKAROUND, using our Piano's built in Audio interface, without having to purchase an external one? Next Subject (but related): I have the Yamaha DGX 670 and record vocals as well. THe only reason to get one with more than two MIc inputs would be to record the piano as well simultaneously (unless recording the pianos own audio interface via one USB to PC and the mic through some simple Solo or 2:2 high quality Interface via another USB to pc (at least 24 bit and 90 some KHZ). But I understand that chances are that those two recordings might be asynchrone /out of sync. I usually record my singing and playing simultaneously but then they are already written/mixed onto one stereo track: Not two separate tracks. ANd noone is perfect so of course I need to overdub anyway. Hence either I have to record the piano first as an audio or midi and then sing but without the mic turned on and then later record the singing while playing back the instrumental track, OR I record both together which often I prefer because then the singing and playing is perfectly synchronised. DO you have any suggestion to my dilemma: 1) SImple 2:2 AUdio interface for only the microphones but not the piano OR 2) Say the Steinberg UR44 (if its not c connection but a, since the pc has only USB a) and then run both the mics and the piano through that one? THe piano has a built in mixer and can handle both the mic and the piano but one at the time to have them on 2 tracks. BUt regarding the vocal effects one prefers to add the effects later when mixing right. It is hard because I like simplicity and don't want to bother with more equipment if it can be avoided. But I bought the Aston Spirit mic and perhaps the quality is kind of let down with the Piano's audio interface (16 bit 44.1 Khz)?? Anyway the latencies are like 11 and 13 ms at 96 in buffer size. Sorry about the length but it's quite complicated.😵‍💫 MY CONCLUSION: I tend to think that IF I need an external audio interface anyway, I MIGHT AS WELL simply buy the Steinberg UR44, although some complain about it. That way I may save the trouble in those technical workarounds. Or I may have to do similar stuff anyway. ANyway with the DSP I need to create a track in CUbase but controle the effects elsewhere. DOES this sound plausible? Or are my latencies adequately small for vocals? I knew yours were 4,3 and 3 in the video but you said vocals would be a problem anyway, despite of them being that small. It just struck me you might be addressing musicians who already own an Audio interface but without DSP. Well mine is inside the piano and the native driver is Yamaha Steinberg USB ASIO. PS: I also tried changing my default CUbase project folder from a regular Hard DIsc to my SSD disk to see if it would lower my LAtency but the values remained the same. I don't know how CUbase calculates the latency. I tried assuming that the default folder and type of Harddrive would be a factor.

[@lorenzojuniormuscoso2547]

Hi Chris, I have a template for recording vocal with all the plugins loaded but off and I still expirience latency. Wy is that if the plugins are off? thanks

[@norctunezmusic]

Hello Chris?I've been experiencing latencies from autotune 9 pro in cubase 10.5 and some waves plugins after mixing and also after I export the mix down cubase doesn't cover them.. The track is a mess.. How can I solve this problem?

[@stix2slivers]

Just a little to fast. I come out of cubase / Focusrite to a Qmix at that point with Qmix I can use external fax. May not be the best set up but it is simple and works but I only get a wash for each player/musician. Any better breakdown I would be happy to learn. Thanks Bruce .

[@TheRobGuard]

I have 3 Audio Interfaces and was wondering if this "latency free" fx like Reverb can apply to all or some of those? These are all Steinberg units: UR22, UR44 and UR824. According to Steinbergs website I guess they are… Oh, I guess they are DSP-based, at least UR44 and UR824.

[@raymurphy5230]

Chris, I am waiting for delivery on the Steinberg ru-rt4. Is the setup of this unit pretty self explanatory with the unit, or are there some educational videos or sites to assist in case one hits that “brick wall”, as has occurred with me often. :-) I am using DP9 DAW Thanks for your videos. -ray

[@mixdownonline]

Hey Ray, check this video, maybe that will help kzbin.info/www/bejne/m6K6Zpute7ikoaM

[@evodem]

Great Tip!

[@mixdownonline]

Thanks for watching!

[@unkletonebeats1956]

hey chris I appreciate your tutorials brother. I have a situation with cubase pro where when recording midi and render in placing to auto events and composing arranging the song begins to drag behind the metronome now when I select the delay constraint compentsation button on the transport the song stays tight locked with the metronome. which would be fine but when i render a part that way the rendered part is much louder that the part with the same instrument that I had done earlier. I notice that the delay contsraint button disables plugins on the master stereo out channel maybe all plugins--is this SITUATION WITH latency? I asl because when I go lower or higher it doesnt matter much.When I track usually buffer at 96 or 128 and mixing and mastering i go 512 or so.

[@unkletonebeats1956]

heres something i had Ik media T-Racks one one my master channel so I disabled it and still had sluggishness , had to totally remove the plug in and timing between metronome and song now perfect weird huh?

[@Dyonysus777]

Hi Chris. I am a singer and a pianist. WOuld you recommend me the Steinberg UR22C (with the DSP effects) or the MOTU M2 or M4 with direct monitoring and the workaround? I don't know if the effects applied with the Steinberg interface can be recorded as well as just being heard via headphones? Others who make videos did not know this. Please tell me. If not recordable then the workaround may be better or simply adding a bit of reverb then more later when editing /processing in CUbase. My preference for the Steinberg naturally owes to using Cubase, so I figured this constellation would mean less potential issues with lack of congruency/compatibility. Thanks on beforehand. I like your videos. 🙂

[@Dan-pq1tp]

Can you do this with FL Studio? I have the PreSonus Studio 26 USB interface.

[@mixdownonline]

Never worked with FL Studio

[@svensvensson8102]

You can send any mixer track in FL to any output on your interface, just disconnect it from master though. If you use direct monitoring on your interface I would keep FX 100% wet.

[@nelsonmoloi3767]

Hi morning Chris so on my side I always use 1 wich is 256 whether iam recording or mixing .or ?

[@Rondozza]

very nice video chris. that's exactly my topic right now. got a new studio with recording room. The monitor situation is not clear yet. I also have a lynx converter. the newer generation. aurora(n) 16 channel usb. Unfortunately there is no control pannel, control software, for the new converters. that's a shame because i'm thinking of buying a new uad converter. after i wait about one and a half year for a software update from lynx. I do not think the brand new interfaces of uad sound much worse than the lynx. the monitoring including effects is much easier in this case. I am torn at the moment. thanks for the video. That gives me another perspective. Greetings from the thomann land :). germany / Nuremberg

[@mixdownonline]

For my Lynx, I had to download the virtual Mixer from their website... Might work for your system as well www.lynxstudio.net/product_detail.asp?i=71 UAD Apollo sound's very good as well, at that level of quality, Lynx and UAD quality converters will be about the same... The new Steinberg AXR4 looks pretty nice too.

[@mikeverzosa2061]

Will this also work for guitar FX? like amp sims or IRs

[@mixdownonline]

No cause an Guita FX an INSERT FX and this worls better using SEND FX like Reverb and Delay

[@mixdownonline]

Hey guys, just to let you know. In the video, I used a CUE MIX to route my reverb out but you don't have to use a CUE MIX do apply this technique, a simple second output will do the same. Create a second output and route the output of the Vocal Reverb (100% wet) Track to OUTPUT 2 and bring up the channel's fader. More suitable if you don't have Cubase or Cubase Pro. Like they say, all roads lead to Rome ;-)

[@neilkendall5499]

Hi, I'm using Cubase Elements 11, so I'll have to do it without CUE MIX. How do I route the output of the Vocal Reverb Track in Cubase? (Still fairly new to Cubase, so I need to know where that setting is for output routing). Thanks.

[@Dyonysus777]

What do you mean with: "More suitable if you don't have Cubase or Cubase Pro." If you don't have Cubase Pro but say CUbase ai 11 then you have CUbase so then what to use. Only CUbase Pro has CUe mix. SO either you have CUbase pro or anything but not CUbase or you are screwed???? Make that clear please.

[@dtsrecordingstudio2431]

Cubase Pro user. Love your videos Chris! If you're not using Cue Mixes within Cubase as headphone sends, how do you communicate with the musicians in your studio? I'm using 4 Steinberg MR816CSXs (32 ch) to record and monitor with. No board. So I have to use talkback in Cubase's Control Room. I know the MR line has direct monitoring, but I just could never figure out how to do it, so I have DM off. Basically, I use a stereo output pair for each Cue Mix to go to my multichannel headphone amp. I also use one Cue mix dedicated to my Behringer P16m monitor mixers. This Cue mix allows the Talkback button in the Control Room to work. (All other channels are sent via Sends to the Behringer's channels) Anyway, It works, but I still confuse myself how I got it to work. I just can't have Direct Monitoring on. As far as the Lynx monitor mixer that you have, there is a similar mixer provided with the MR interfaces that has faders and such. However, you can't have the MR Editor mixer open the same time as Cubase....for whatever reason. This make any sense to you? Thanks! - Dan from New Jersey

[@mixdownonline]

Hey Dan, I use telepathy to communicate...lol..seriously CR is a great way to set up a talkback. I used to have my Presonus Central station taking care of my talkback situation and it worked well until it broke off :-( Now, I use a Shure 55SH Mic for talkback using one of my preamps that I balance through the Lynx Mixer to the monitor mixes. Now the only reason I do it this way is that this mic has an on/off switch and I find it very convenient to use as a talkback because of that switch. I would love to help out with your MR talkback but I don't know the MR line, unfortunately. If your talkback setup works well, keep it this way!

[@dtsrecordingstudio2431]

@@mixdownonline Yup! I agree. If whatever way works for you....stick with it. Engineering is all about finding a path that works and makes sense to you that does the job! Great videos, keep 'em coming! :)

[@MrTerryGene]

I use Cubase 6.5. I have been doing that with the Saffire Pro40(headphone mix). However, how can I not only "hear" the Effects, but "record" the effects on the way in using "inserts" on the "Input" channel? Is there also a way to accomplish that?

[@mixdownonline]

Yes if you Insert your reverb plugin in the INPUT channel in Cubase, that will record the effect

[@nathanponzar3816]

Is there any way to do this with compression? It seems like you're still going to get a doubling sound, because the compressed and non compressed signals are not really "wet and dry" in the same way reverb is. I like monitoring vocals and acoustic guitar with a fair amount of compression, just feels better. I used to do this with software monitoring and a firewire interface and had virtually no latency noticeable at all, but now I'm using a USB interface and I notice the delay big time. Do you think a thunderbolt interface would circumvent this?

[@mixdownonline]

Unfortunately, that trick works best with effects like Delay and Reverb, compression will not work, you will hear the latency

[@nathanponzar3816]

@@mixdownonline That's what I was thinkin. Any other workarounds you can think of? I'm thinking maybe buy a small mixer with built in compression and routing the playback and interface input signal out into the mixing board, then out to headphones. Alternatively, I may just buy a thunderbolt interface and use software monitoring. This is a good trick you mentioned here though for verb!

[@martynauto8089]

But what if you want to monitor with insert effects ?

[@mixdownonline]

You can by not activating Direct Monitoring and Monitor from Cubase, if you don't mind about a bit of latency. When recording bass, guitars and vocals, a bit of latency won't bother much.

[@diegoalejo15]

you can enable "Constrain delay compensation"

[@luiscepeda3058]

is it possible to do that with an UR22?

[@JamDauntlessTurkey]

Wow!

[@kromanoidx]

Sorry for the stupid question but, if I have inserts creating some monitoring latency the input latency is not being affected, right?

[@mixdownonline]

If you monitor your Input from your sound interface bypassing Cubase, all plugin latency will not affect that signal :-)

[@Tbemod]

So I have to toggle on the "monitoring" on the additional audio track for me to hear the effects in the cue mix?

[@mixdownonline]

Yes, if you want to use this trick to bypass latency

[@Tbemod]

Chris Selim - Mixdown Online got it. I will try it, mb throwing more effects like compression etc is possible :)

[@Tbemod]

@@mixdownonline I tried your method and it worked, excellent job! Now it's really easy to record with plug-ins and get only a dry signal. Thank you! Although I have a question, mb it's worth another video. In cubase, in VST Audio System Advanced settings, what do ASIO-GUARD and the options below it do? I guess it's some latency compensation whatnot. I'm just recording vocals to a 2-track beat. What does the audio priority do ?

[@joeybaronstudio]

Can’t the lynx send and effects buss out? And just use a effects unit? You can pick up a lexicon 4 channel unit for like 80$ .. I have an old i7 system and motus and can run 16 buffers all day long when tracking and I use external effects thru cubase and kick up buffers when mixing, but I do have a digital mixer, I do run into buffer problems when using midi vsti sounds, I would love to try an new i9 system

[@mixdownonline]

Yes, you can use external FX with the Lynx. It is definatelly a good alternative for monitoring with fx... I'm building a new PC with an i9 9900k processor...I'l make a video on this

[@joeybaronstudio]

Chris Selim - Mixdown Online I’ve been waiting to see an i9 video, I cannot believe that it’s so much faster than the i7 ... my i7 system is the fastest computer I’ve ever seen

[@Bring_MeSunshine]

Popped by the channel and did a search on Delay Compensation. There is some facility to work with DC in Cubase; would you consider covering it in a vid, please?

[@mixdownonline]

Yeah, sure...I will add this topic to my list :-)

[@Bring_MeSunshine]

@@mixdownonline Thank you, Chris. I'm determined to go deeper into Cubase, but there's so much functionality worth getting your head round that it's great to get input from a user with experience

[@kris_lx]

That's pretty clever, why didn't I think of that? 😎

[@mixdownonline]

You're welcome

[@halidharis]

what about mixing stage..how do i fix latency after i insert a plugin..thanks for your answer..

[@mixdownonline]

Latency doesn't matter in Mixing because you're not dealing with a direct sound being recorded

[@norctunezmusic]

I have been having same problem from plugins like autotune 9 pro, latencies of upto 83ms... How can I solve this too while mixing?

[@YA-sound]

Norctunez Music if you’re mixing you’re not recording.. all audio is getting the same latency when you play it back so it doesn’t matter. If certain tracks were already recorded with latency because of the plugins; I’m sorry but you just gotta fix this by hand now xD

[@pianolover5417]

I'm gonna do that too! 😁

[@sub4037]

Hey Chris, a lil of topic but I was wondering if there is a way to convert a cubase project from 440Hrz to 432Hrz? thanks man...SUB403

[@mixdownonline]

Hey SUB403, 440Hrz is the frequency of the A note so when recording or playing in 432Hrz that means that you detune your instrument a bit until the A note (the reference note) hits 432Hrz. In some cases, 442Hrz will be used in Orchestral genre and 432Hrz in rock to get a tone down feel. Mainly used for ''Live'' purposes...All that to say that this is not a DAW project setting that you can change for an entire project. It has to do on how you record your instruments :-)

[@123robbyd]

I've played around a bit with 432 tuning in Cubase PRO....and Chris is right. My solution was to take my finished 440 mix, load it into a new "mastering" project, and then use pitch shift to take it down about 32 cents using the maximum settings. Cubase 8 did this kinda OK. Cubase 10's pitch shift algorithms are vastly improved and it does an amazing job. BTW: C-10's new Vari-Audio ("auto-tune" functions) are killer. I use the manual controls 98% of the time. It takes a little time to learn how to use it transparently, but it's worth the effort.

[@OriginalRaveParty]

For clarification, this method is useful for send effects which naturally include delay-like processing (reverb, delay). It's not useful for insert effects, or effects chains (EQ, compression, de-essing, widening, exciters, or some uses of chorus/flange/phaser). The only way to hear yourself with an awesome effects chain is outboard gear. That's one of the main reasons why outboard gear is still so necessary for professional work.

[@mixdownonline]

That's exactly it. Thatis how I use this. agee 100%

[@davidtrueheroakad-true8318]

Would a focusrite Scarlett solo interface solve the issue of latency while using auto tune effect ?

[@OriginalRaveParty]

@@davidtrueheroakad-true8318 There are live, low latency vocal tuners available, but any software which heavily processes sound, like auto tune, causes latency. It's not the fault of the audio interface. Also, Focusrite's audio drivers are crap despite their gear being great. I own an 18i20 and I have to use ASIO4ALL drivers to avoid recording pops and clicks.

[@samickjp]

Can you explain?

[@OriginalRaveParty]

@@samickjp Musical plug-ins are like a calculator. You put raw audio in at the start, they perform a complex process and then send out the results, such as the audio with the eq or compression applied. The process takes up several milliseconds, which obviously means that the more effects you have on the raw audio, the more delay is added. If you are playing an instrument into that chain of effects, even 40 milliseconds of delay can really start to mess up your playing and performance.

[@enricmarshal]

Excelent explanation. However I don't quite understand what cue mixer are so if you could make a tutorial about it, it would be great. Thanks my friend.

[@mixdownonline]

I actually covered Cues Mixes in my course theultimateguidetocubase.com But you don't have to use a Cue Mix for this technique to work. Just create a second Stereo Output in Cubase and Route the output of the Vocal Reverb track to that Output. In this case, opposed to my video, bring the Vocal Reverb track fader up.

[@enricmarshal]

@@mixdownonline ok, thanks Chris

[@davidtrueheroakad-true8318]

What about auto tune effect? Would the same method work ?

[@mixdownonline]

If a tiny bit of latency doesn't bother you, you can. I would apply the Auto-Tune effect after the track is recorded if I were you.

[@edisonceni1092]

What about "asio time overload" issue? It keeps happening after a little time using cubase monitor in a prearranged cubase project for live situation? It is just an amp/rig modeler inserted but it keeps popping ang crackling expecially on the delays and reverbs? The high buffer size wont help in a live situation because of latency

[@mixdownonline]

The sound interface and especially the CPU speed of a computer will have an influence with the handling of the buffer size.

[@backspin6698]

Great vid Chris, little technical but we can watch it many times :) My computer is also 6 years old. After I got a SSD hard drive(can recommend that) I can use 128 in buffer all the time (my projects aren´t that big either). Love to not have all the pops and crackles. And I don't mind singing with a little latency.

[@mixdownonline]

128 works well for me too!

[@imsoundtrack]

Hi ...!!!i also use a reverb...but i put the vocal output channel to -no bus- and then i just go to my reverb and i drag the (dry) button completely down and the (wet) button to 0 or less...when i am done i put the output channel to the correct bus again....it works perfectly!!!!! Zero letancy!!!!!!!

[@mixdownonline]

That will work...All roads lead to Rome ;-)

[@cgb1394]

i'm new and watched this a few times..still don't get it! I use Ableton Live and even a little latency recording guitar throws me off.

[@BURCO2010]

Is there any one has cubase studio in London

[@reganjo1955]

I may be missing something but this doesn’t make sense to me. I don’t see how fader down on essentially an aux track for Vox doesn’t incur input monitoring latency. I get direct from mic to tascam (5/6) I get 2048 samples created huge lag when monitoring. I understand you are sending Vox aux to cue mix (3/4) and how Cubase is not processing (you must be pre-fader) I don’t understand how the mic input audio sent to Vox aux cue mix doesn’t incur latency. Cubase is processing that audio through little plate and this MUST be hitting the DA on the way out through ASIO. What am I missing? When I get time I will try to reproduce with my babyface pro and Cubase. I will try different latencies starting at 64 and going up to 2048 to see if latency creeps back end.

[@mixdownonline]

I monitor the Vocal Direct Signal from the Sound Interface. What I monitor from Cubase is only the effect (Reverb). That why my Cubase vocal channel is way down but send ig that signal pre-fader to the reverb. The Reverb will have latency which I don't care cause it's a reverb but the Vocal is monitored strait from the sound interface. Hope that helps!

[@philippmindrin3202]

That was a good tip, but what if it comes to use an autotune?

[@korkenknopfus]

The treatment of latency in guitars and amp emulations is clear and relative simple, but I hadn’t really thought in this possibility that you explain with voices. Very interesting. To be honest, I am still not very sure if I understood well. But I can think the following with my system, so like a simple version of what you have: I have a Presonus 44VSL where I can control with a hardware mix fader the amount of direct hardware and software monitoring. If I would record vocals, I think I could set the monitor mix fader midways and to monitor the dry voice directly (hardware) and via software to monitor only a FX channel with reverb, for example, without dry voice. If I understood good your explanation, it should be something like that, but without using cue mixes. Thank you again and good that you are back. And guys, check out the Cubase course of Chris, because you will learn a lot.

[@mixdownonline]

You got it, that's how you would do it with your interface. Happy you like my Cubase course :-) Thanks!

[@korkenknopfus]

Good then, I will try that. Man, after one read it, it sounds so simple, but I really had never figured it out. And... yes, I like the course, not the “curse”, haha, I edited it. In Spanish is “curso”. Curse is simply “maldición”.

[@mixdownonline]

😂 Don't worry, I understood what you meant

[@korkenknopfus]

I forgot to mention too that the Audibox VSL series (44VSL, 22VSL and others) had a Presonus software called VSL, to do something like that is discussed here. There was some latency, but much smaller that the one using the DAW. The VSL software had compressors, eqs and reverbs, thought to help in recording (for example to make it comfortabler to a singer). But the software was discontinued on 2015. I haven’t checked out if there are options yet.

[@mixdownonline]

Thanks for the info!

[@seppy3272]

will this work with fl studio?

[@mixdownonline]

Never worked with FL Studio

[@rayerscarpensael2300]

I've never listened to guitars via cubase for this reason, only the click via my headphones, listening to the live recording at the amp,, the latency is just way to bad to be able to play on time with plug ins or without; But on the other hand i did not understand and still do not know what a 148 bufferzone is???

[@mixdownonline]

The buffer size will affect the amount of latency your DAW will generate...Lower buffer = Less latency but requires more CPU resources... Higher buffer = Requires lower CPU resources but will add higher latency. Hope that helps!

[@TheRobGuard]

@@mixdownonline I Think I understand this well enough to keep latency low when recording a guitar using for example AmpliTube 4, Guitar Rig Pro 5, etc. So I try to not go over 5 ms latency, basically putting it on the lowest buffer size. Still the sound is not crackling, etc. Then when mixing I raise the buffer size a lot and can add a lot of plugins, etc. But… What is there a payoff here? When using a very low buffer size? Is the sound worse in any way? I'll can add that I dont ever use Direct monitoring...

[@mauricioapoteom2604]

How to fix the latency on a bus?

[@datamasked623]

To be clear, analog equipment has no latency. Latancy only occurs between the audio interface and computer CPU -- it's the time it takes to process. The DELAY you notice when you send audio out to a compressor or mic pre in your rack or something is the physical roundtrip. That ain't latency. The closer your gear is physically located to where you're sitting, and the shorter the physical cabling is connecting it to your interface, necessarily shortens the roundtrip time it gets sent out and comes back. I hear a lot of people use "latency" and "delay" synonymously. They are not the same thing. Latency ONLY occurs in the digital domain. Calculating the round trip delay can be determined by your sample rate. For instance, if you're at 44.1k and you have a delay of 100 samples, the formula is 100 samples/44,100 samples x 1000. That'll give you the compensation delay, in milliseconds, you need to set your DAW's hardware inserts to. I know that's not really the point of this video necessarily, but a lot of people may be put off outboard gear because they don't want to deal with that, but it's really easy.

[@orientmob]

Yo Chris from where are your father from ? Greetings

[@isekaiguy69]

Didn’t have issues recording vocals with a buffer size at 512. latency was at 12.5 This was with the UR-RT2 with never transformer on.

[@mixdownonline]

Did you monitor using the DSP MIX FX? If so, latency won't matter cause you're bypassing your DAW for monitoring :-)

[@isekaiguy69]

Yup! The magic of DSP 😁

[@quantumsound2423]

Presonus Studio One has the best low latency monitoring mode. Logic Pro X’s low latency mode comes very close to studio one. But cubase is far behind.

[@TheHirade]

Latency is a matter of the audio interface, not the DAW, so your comment makes no sense The 11 thumbs up are weird...

[@streetlogic]

yeah I set buffer to 96 and is perfect with effects on but my laptop can handle it so guess im lucky

[@mixdownonline]

Cool stuff :-)

[@Thermolizer]

What laptop do you have?

[@eltitonthemusic]

hmnmnm............. pretty sure thats not what i was looking for but thanks for nice explanation

[@mixdownonline]

You're welcome

[@ronanzann4851]

It depends on what you mean by "no latency".

[@nikolaosunikgiannulidis8800]

yeah this is nice, but it only works of your actual project in your daw is not taxing your cpu too much. as soon as you have 50+ instrument tracks and 100 uad plugins your latency will go up 10 fold and software monitoring won't work anymore. one of the main reasons people still use dsp based interfaces.

[@mixdownonline]

Actually, if like I did in the video you use software monitoring to monitor the Reverb only, it will work well even if you increase the latency to a 2048 buffer. Latency on a reverb doesn't matter much as far as you have your Dry vocal without latency like in the video. Try it out!

[@ralfnolte174]

Umm, just freeze most of the already recorded tracks as long as you do the recording. this way, instruments and plufgins are of no concern. For 50 tracks, i would even generate a few stems, (groups->render in place) for most of them, then disable the original tracks and bob's you uncle.

[@StaticTeaz]

so "no" = 128ms?

[@mixdownonline]

The direct sound has no latency only the FX (Reverb), that's the workaround

[@StaticTeaz]

@@mixdownonline But the title is "recording WITH FX with no latency"... That implies one has a processed signal-- that is, can track with effects--in real time. What is the workaround? I still don't understand what you're saying with this.

[@matenorth]

A bit of clickbait here. Its either DSP, dry direct monitoring or latency through DAW. There is no other way.

[@matthewward5813]

Do you know how a reverb works? A lot of times reverbs are set to 40 or 50 ms of predelay anyways. All I do is add an extra track and add a reverb plugin to it. Arm it to record with monitoring through the daw turned on. At the same time have monitoring turned off of the track you are going to record the dry vocal to. So through your headphones you will hear the dry signal that comes striaght from your interfaces direct monitoring, while hearing the reverb that is running through the daw with latency of say 15 ms. All that would sound like is having a reverb on your original signal that has the reverb predelay set to 15 ms.

[@syedshahbazali1417]

Skip to 9:38 thank me later

[@only4posting]

4 ms at 128 2 ms at 64 ~1 ms. at 32 At 48khz If you use 96khz, it might go under 1ms If you use 192khz, it might be even lower. Of course, at 32 samples, and 192/24, any small-medium project would destroy even the most powerful computer. But what if you bounce just a couple of tracks... and then create a new project, set the card to 32 samples and 192/24, import/convert the bounced files, and start recording ? Any computer should be able to record a few tracks, while playing 5 or even 10 audio tracks, all at 32 samples, and 192/24 ? Have you tried to create an empty project, set 32 buffers, 192khz and 24 bit, and start recording some vocals, 'JUST' to see how it performs and sounds, under 1ms of latency ? 1ms, that's super low.... And I'm sure most mixers, that claim having 'zero latency', STILL have some latency. 0.5ms.... 0.4.... 0.3..... but 0.0...... i seriously doubt it.

[@mixdownonline]

Hey bro! I did not try this for the simple reason that I like to monitor directly from my Lynx with no latency and for me, even if the idea is good bouncing tracks around in another project to record at 32 samples will just add too many steps in my workflow. Thanks for your comment on this bud :-)