Jump to content

recording with Autotune and input monitoring


Jordi

Recommended Posts

I hate to say this but that's the cart before the horse. Unless you are trying to do this for a live performance I see no benefit in working this way. I see many reasons not to. 

You won't hear any effects in real time without turning on input echo. Direct monitoring is just that. It is the signal from the mike input directly to your headphones or monitors. You will only hear any DAW processing at the output stage. 

 Running an auto tune plug is is a CPU intensive task. On most computers this requires higher buffer settings so there are no dropouts of glitches.  To run at a low buffer setting you will then need to either have the worlds most powerful computer and something like an RME audio interface.  A so, so  computer and a so, so ASIO driver will require  a higher buffer setting which equals more latency resulting in an echo in the headphones. 

If you want auto tune in real time then look into a TC Voice Live processor. https://www.tc-helicon.com/series.html?category=R-TCHELICON-VOICELIVESERIES

  • Like 1
Link to comment
Share on other sites

8 minutes ago, JohnnyV said:

If you want auto tune in real time then look into a TC Voice Live processor. https://www.tc-helicon.com/series.html?category=R-TCHELICON-VOICELIVESERIES

+1 -- i know a few people who use the vocal extreme 3 w/ backing tracks and live instruments (guitar, keys) + harmonizing (which is not trivial to do even with the auto-pitch stuff) with great success and really excellent live shows.

Link to comment
Share on other sites

16 minutes ago, JohnnyV said:

I hate to say this but that's the cart before the horse. Unless you are trying to do this for a live performance I see no benefit in working this way. I see many reasons not to. 

You won't hear any effects in real time without turning on input echo. Direct monitoring is just that. It is the signal from the mike input directly to your headphones or monitors. You will only hear any DAW processing at the output stage. 

 Running an auto tune plug is is a CPU intensive task. On most computers this requires higher buffer settings so there are no dropouts of glitches.  To run at a low buffer setting you will then need to either have the worlds most powerful computer and something like an RME audio interface.  A so, so  computer and a so, so ASIO driver will require  a higher buffer setting which equals more latency resulting in an echo in the headphones. 

If you want auto tune in real time then look into a TC Voice Live processor. https://www.tc-helicon.com/series.html?category=R-TCHELICON-VOICELIVESERIES

Well . . .  that is why you route the input echo channel to another audio track to record it with the effect.

Most rappers an even singers prefer this way of recording. My system handles this with absolutely no issue if you do your setting correctly as i have said above. 

Link to comment
Share on other sites

3 hours ago, JohnnyV said:

I hate to say this but that's the cart before the horse. Unless you are trying to do this for a live performance I see no benefit in working this way. I see many reasons not to. 

You won't hear any effects in real time without turning on input echo. Direct monitoring is just that. It is the signal from the mike input directly to your headphones or monitors. You will only hear any DAW processing at the output stage. 

 Running an auto tune plug is is a CPU intensive task. On most computers this requires higher buffer settings so there are no dropouts of glitches.  To run at a low buffer setting you will then need to either have the worlds most powerful computer and something like an RME audio interface.  A so, so  computer and a so, so ASIO driver will require  a higher buffer setting which equals more latency resulting in an echo in the headphones. 

If you want auto tune in real time then look into a TC Voice Live processor. https://www.tc-helicon.com/series.html?category=R-TCHELICON-VOICELIVESERIES

I used the older TC-Helicon VoicePrism Plus live for years - not for auto-tune, but for the voice modelling & feedback destroyer.  There was a setting that would make normal singing sound as if you were belting it out... fantastic for a 3 hour gig and being able to still speak afterwards!

It could also add rasp to your vocals, and the weird thing was by practicing with the rasp on, I actually managed to develop a natural rasp myself... I wonder if that'd work for tuning too?

I later upgraded to the VoiceWorks Plus, which has improved voice modelling... but both the standard "Voice Prism" and "VoiceWorks" will do autotune... as will other boxes like the DigiTech vocalist series.  They're all probably discontinued now, but you should be able to get used ones on eBay/reverb.

  • Like 1
  • Great Idea 1
Link to comment
Share on other sites

I only bought my Voice live play acoustic because I saw another solo act in a club one night and his vocals just blew me away. I went up to him on a break to ask.  Bought one right away. 
It is more than just a harmonies generator. That alone is worth it and not only harmonies it can blend a pitch correction in with the dry.  


First it solved the issue I had with my mixer not having very good  effects for vocals. Not only are the effects good quality but each patch has its own so I get what I want with a foot switch. 
It is an amazing processor for my acoustic guitar as well. That is global but I supplement with a zoom pedal. 
It’s my acoustic guitar DI as it has XLR output for that. And it got to be the fastest guitar tuner I’ve ever used. 
Then it’s also my in ear monitors mixer. I can not only blend my vocals and guitar but it has a AUX input that I can patch a send from any mixer into. 
So if I take it to shows where I’m using a house system or there’s a guy running the board I still have control over my in ear mix. 

Vocal harmonies, vocal pitch correction, vocal effects, guitar DI Guitar effects, Guitar Tuning , in ear monitor mixer with Vocals, Guitar and Aux from mixer. 

Link to comment
Share on other sites

19 hours ago, Will. said:

I sometimes record rappers in CbB. 

Make sure your sample rates are in sync'd. 

Recording bit deph should match that of your Asio>Daw>Autotune.

What mean? :

Make sure your sample rates are in sync'd. 

Recording bit deph should match that of your Asio>Daw>Autotune.

My project is: 48kz 24 bits, autotune don't have the possibility to configure de sample rate or bit depht, it work with the sample rate and depth bit of the project.

Link to comment
Share on other sites

15 hours ago, JohnnyV said:

I hate to say this but that's the cart before the horse. Unless you are trying to do this for a live performance I see no benefit in working this way. I see many reasons not to. 

You won't hear any effects in real time without turning on input echo. Direct monitoring is just that. It is the signal from the mike input directly to your headphones or monitors. You will only hear any DAW processing at the output stage. 

 Running an auto tune plug is is a CPU intensive task. On most computers this requires higher buffer settings so there are no dropouts of glitches.  To run at a low buffer setting you will then need to either have the worlds most powerful computer and something like an RME audio interface.  A so, so  computer and a so, so ASIO driver will require  a higher buffer setting which equals more latency resulting in an echo in the headphones. 

If you want auto tune in real time then look into a TC Voice Live processor. https://www.tc-helicon.com/series.html?category=R-TCHELICON-VOICELIVESERIES

Thanks your reply, i know all limitations about real time effects and latency.

The reason to work with autotune in real time in a track recording a vocal is , all reggetton singers use autotune like cher style effect to sing while recording their voice.

they usually do it in fruity loops without latency, I would like to be able to do it on my Daw: Cakewalk.

I know that reggetton is not good music, but it's work, haha, please don't make comments like this to me, haha

Link to comment
Share on other sites

2 hours ago, Jordi said:

What you mean?

Recording bit deph should match that of your Asio>Daw>Autotune.

Your Asio needs to match that of your DAW which than goes in to Autotune. 

So, your buffer size needs to match. You Asio buffer size, You DAW buffer size and your buffer size in Autotune. 

This mean: If your Asio driver is the only driver selected in Windows as the primary drivers - set that sample rate according to your DAW which would than go into Autotune. What you're running in your DAW needs to match that in your O.S. 

From there you can adjust and match the buffer size in both your Asio drviers and in Autotune - depending on your Audio Insterface drivers these can be set independently. For instance with Focusrite Asio drivers whatever you change in the DAW change in windows too and also true the other way around. 

RTM of autotune. Try to disble that annoying Windows Defender too. Mine is permanently disabled. 

PS: I have no issue in regard with recording live vocals with Autotune with these settings. 

Link to comment
Share on other sites

3 hours ago, Will. said:

Your Asio needs to match that of your DAW which than goes in to Autotune. 

So, your buffer size needs to match. You Asio buffer size, You DAW buffer size and your buffer size in Autotune. 

This mean: If your Asio driver is the only driver selected in Windows as the primary drivers - set that sample rate according to your DAW which would than go into Autotune. What you're running in your DAW needs to match that in your O.S. 

From there you can adjust and match the buffer size in both your Asio drviers and in Autotune - depending on your Audio Insterface drivers these can be set independently. For instance with Focusrite Asio drivers whatever you change in the DAW change in windows too and also true the other way around. 

RTM of autotune. Try to disble that annoying Windows Defender too. Mine is permanently disabled. 

PS: I have no issue in regard with recording live vocals with Autotune with these settings. 

Ok, I understand what you are explaining to me, but in the Windows 10 sound settings from the Windows system settings, I cannot select the Asio drivers, I have a Motu Ultralite mk3 Hybrid.
Windows only accepts audiowave drivers, at no time does it show me Asio drivers.
What I can configure is the sample rate and bit depth that only lets me set it to 48kh and 16 bits.

Of course in Cakewalk i use Asio drivers.

Link to comment
Share on other sites

1 hour ago, Xoo said:

That's correct - Windows sound settings doesn't let you select ASIO; Cakewalk (or a.n.other DAW) does.

Mine does. 😐 My focusrite 2i2 is my only device on my system as my main soundcard, there for, I have access to my Asio Drivers. So whatever I change in the DAW affects my Asio Drivers for my interface (aka system drivers.) 

Edited by Will.
Link to comment
Share on other sites

1 hour ago, Will. said:

Mine does. 😐 My focusrite 2i2 is my only device on my system as my main soundcard, there for, I have access to my Asio Drivers. So whatever I change in the DAW affects my Asio Drivers for my interface (aka system drivers.) 

Could you send a capture screen of this please?

Link to comment
Share on other sites

There is no settings for ASIO in Windows. You install the driver and if you disable all other audio devices in Windows settings/ Manage devices then you are using your interface and ASIO.  
Open Cakewalk audio settings and you can confirm that.
Interfaces these days are 24 bit and that is not adjusted in Windows.  

Open the Motu control panel to change clock rate and buffer setting.  

 

Edited by JohnnyV
Link to comment
Share on other sites

15 hours ago, JohnnyV said:

 if you disable all other audio devices in Windows settings/ Manage devices then you are using your interface and ASIO.  

. . . . and that is the answer. 

What I have been saying the whole time. 

Link to comment
Share on other sites

In a way yes but you said 

On 6/22/2023 at 4:53 AM, Will. said:

This mean: If your Asio driver is the only driver selected in Windows

That actually confused me. I think you made a typo and meant to say ASIO device.  Not driver. The op was looking for a setting that doesn’t exist 🫤
Since the beginning of time many of us have always disabled the computers audio system to avoid conflicts.  On my main DAW I didn’t even install the driver for it when I rebuild it. But windows update screwed that up one day. 
Generally the majority of people simply install the ASIO drivers and never even have to think about anything else.  
And even thought it probably doesn’t matter I always set The sample and bit depth in Windows to match my interfaces settings. Just  as you are advising. That  seems to work for me when sharing other apps with Cakewalk running.  

Link to comment
Share on other sites

Please sign in to comment

You will be able to leave a comment after signing in



Sign In Now
×
×
  • Create New...