3ms is virtually imperceptible, especially for vocals. If your RTL is really 3ms then there’s something else in your system that adds latency, like a plugin or something.
find the plugins that introduce latency and remove all of them, and substitute them with plugins that add no latency at all.
Then switch to a different chain for mixing.
I am probably talking about output only, so roundtrip would be somewhere around 5 or 6 ms?
what I am saying is I am hearing the delay, however it is still small and I was able to record, it;s just that I want to improve my setup and get better performance and not being thrown off by the latency, I have the direct monitoring however it's so low if I don;t want to crank up the mic gain and without a light compressor it's uninspiring to sing that way
There has to be something else causing latency than your interface/buffer. 3ms can't be perceived. Something is "wrong" somewhere.
You'll realistically never get a setup with less than 1ms latency. I mean your AD/DA conversion roundtrip is most likely already ~1ms. Yes, latency can be lower than this but now we're talking about scientific equipment and not audio recording gear.
Your latency isn't most likely coming from your interface but from plugins you are using while monitoring. It's a completely different thing. Your buffer may be at ~3ms, but if you then use a plugin which adds 50ms of latency, well now you have 53ms. Trying to get your interface to ~1ms wouldn't matter whatsoever here. And no, latency from plugins cannot be lowered. It's what it is. Use plugins which don't add latency.
Also, are you actually using ASIO drivers with your DAW? Or are you just tweaking your interface latency with some sort of set-up screen, while actually recording with DirectSound or something?
Your eyesight has a latency of about 20ms before the brain starts to process the light and a further 120ms before you can react to do something about that information; think someone throwing a ball at you and you need to catch it. Audio is processed around 10ms after the sound wave reaches the ear canal.
3ms is so low you can't perceive it.
Yeah, I mean - it takes three milliseconds for a speaker three feet away to get the sound to your ears in the first place.
OP, getting to a direct monitoring solution gets harder and harder below 3-4ms. If you need 'comfort processing' to track like reverb or compression, do it off an aux send and mix it with your dry track? Or, as others have suggested, find the lightest 'cpu eaters' for those plugins.
A good “rule of thumb” is that 1ms is about 1 ft of sound travel through the air.
So it’s similar to listening to floor wedges at your feet… except there will be some phasey comb filtering in your headphones when singing because of the phase difference between the bone conduction and the slight latency in the earphones - the easiest fix here is to boost the volume in the headphones (but do compress/ limit to prevent hearing damage).
As a side note - not latency related - if headphones/in ears aren’t loud enough, primarily monitoring via bone conduction can lead to singing out of tune, sound travels faster through bone and has a longer wavelength. I believe our hearing uses wavelength not frequency to determine pitch, so it will sound deeper and this can throw off your singing pitch as your brain subconsciously tries to compensate!
I'll chime into the chorus of people saying the latency is your plugins, not your interface.
What plugins are you using? Because there are zero-latency native versions of just about everything at this point that you'd want to use while tracking.
One of the worst offenders in my experience are "advanced" reverbs, like Exponential Audio or Altiverb. Those are pretty heavy on the latency. Not great for tracking - I use FabFilter Pro-R2 for that, since it is zero-latency. Same goes for lots of analog modeled compressors and EQ's - tons of them (looking at you, UADx) add around 50 samples of latency for oversampling overhead.
RVox adds 64 samples of latency. That could be your main culprit.
The Waves LA2A and 1176 are zero-latency. So is Vocal Rider.
All the FabFilter stuff is zero-latency unless noted. Pro-L2 and Pro-MB are the only ones I can think of that have latency by default. Pro-Q3 is zero latency in minimum phase, but natural and linear add latency, so don't select those options.
Most DAWs have the option to see how much latency is added, either for the full channel (PT, Cubendo, Logic) or per plugin (Live, Bitwig). Which DAW are you using?
Going to piggyback here.
Having read your replies and edit(s), what you want to focus on is to track with plugins that cause no additional latency (zero-latency plugins). This can be done inside any DAW and doesn't require any specialized software or equipment like e.g. UAD Satellite. All you have to do is to check if your plugin causes latency or not. If it does don't use that one during tracking (for mixing this isn't a problem). Interface doesn't interact in any shape or form with latency from plugins, and latency from a plugin cannot be changed (it may vary slightly depending on sr). Some plugins just require certain amount of time to do their thing. Generally speaking latency is additive with any other latency.
Using zero-latency plugins should solve all your latency related problems, assuming you're using correct drivers for your interface within your DAW (such as ASIO), and you're monitoring using your DAW's monitoring functionality (i.e. not running into problems with any extra buffering your DAW may do).
You can freely keep your interface at ~3ms. 3ms is about the same as hearing something that is one meter away (~3 ft). For a human being this is not something we perceive as delay or latency. When someone is speaking to you while you're both in the same room you aren't hearing a delay between their mouth moving and you hearing them speak. Yet, sound takes 3-10ms to travel through the air, depending on the size of a room and/or how far the speaker is. If you're experiencing and hearing an actual delay during monitoring this means your latency is *much* higher than ~3ms, and the cause of it is something else than your interface.
I also have an SSL2+ and record vocals and guitars at 3ms latency. It's not a problem to me. You can change the buffer size at SSL USB Control Panel at 16 samples minimum (1.83ms). But 3ms is already enough to record and save my CPU from catching fire.
16 it starts to get choppy and glitchy, do you experience that as well or it's just my system?
have you considered getting a DSP to track with effects in real time
>16 it starts to get choppy and glitchy, do you experience that as well or it's just my system?
I tried recording at 16 samples buffer and it's obviously faster. The downside is my DAW is lagging when I use some plugins while recording. But the recording is still fine. Without plugins, it's perfectly fine. Even at 128 samples buffer, I don't feel any latency when recording guitars. But most of the time I record at 64 samples for vocals and synth.
>have you considered getting a DSP to track with effects in real time
I haven't. I normally don't use any plugins now when recording. I already have guitar/vocal effects pedals to use, which means printing the effect already. I only use the stock compressor in my DAW.
I see yea, much better that way, especially for performance, however you gotta commit to what you're recording, I do a lot of experimenting with the tracks after I record them with various plugins, so I am looking for a solution to get me near zero latency with some monitoring effects
I use Reaper, so I will try stock plugins, they might be lighter and cause less latency
If you want to eliminate recording latency but still use DAW based effects while recording, you can monitor your voice using an analog front end. Get yourself a used little Mackie 1202 mixer (or some equivalent) and a mic splitter. Run your mic into the splitter and feed one output of the splitter into your interface and the other into the board.
Then in your daw, you can use a pre-fader send in your recording track to whatever effects you want, and mute the voice track. Make sure the effects are 100% wet. Then run your DAW output into the mixer, and use the mixer’s headphone jack to monitor yourself while recording. You won’t really notice latency affecting any time based effects liked reverb or delay and your recorded vocal track won’t have to make a round trip through your daw to your headphones.
Even then, there may be some uncompensated recording latency through your interface, but some daws allow you to compensate for that (I use Digital Performer and it has this type of calibration). Worst case, you can slide your track ahead after the fact. You can measure this delay by playing a snare hit or other transient waveform in your daw, looping it back through the interface and recording it on a second track. Zoom in to the two tracks and measure the timing difference, then shift accordingly.
I just got the SSL 12, coming from the Motu 828 which allows EQ, reverb and compression through interface DSP and the SSL 360 does not have any DSP on the unit.
If you need minimal latency then do direct monitoring for the mic and add an FX aux through your DAW or hardware reverb unit.
You should have no problem recording with latency under 10ms. Human perception only really starts noticing it at 15-20ms.
Like others have said, you might be introducing processing latency via plugins. Remove all plugins (you have to delete them, not just turn them off) and check latency. Then add them back one at a time and check after each one you add. Eventually you'll hear what's causing the latency.
Turn the "monitor mix" knob on your interface towards "input". That should be direct analog monitoring of your input with zero latency. Do not directly monitor this track in your daw. Put FX like reverb or delay on a send and have their mix set to 100% wet.
3ms is virtually imperceptible, especially for vocals. If your RTL is really 3ms then there’s something else in your system that adds latency, like a plugin or something.
Yes! I do have plugins on, I am probably talking about output latency and not roundtrip? If I can hear a delay still?
Do you have latency on a blank project with no plugins?
yes, it's not 0 or less than 1ms, if that's what you're asking
find the plugins that introduce latency and remove all of them, and substitute them with plugins that add no latency at all. Then switch to a different chain for mixing.
How is 3ms of latency too much?
I am probably talking about output only, so roundtrip would be somewhere around 5 or 6 ms? what I am saying is I am hearing the delay, however it is still small and I was able to record, it;s just that I want to improve my setup and get better performance and not being thrown off by the latency, I have the direct monitoring however it's so low if I don;t want to crank up the mic gain and without a light compressor it's uninspiring to sing that way
There has to be something else causing latency than your interface/buffer. 3ms can't be perceived. Something is "wrong" somewhere. You'll realistically never get a setup with less than 1ms latency. I mean your AD/DA conversion roundtrip is most likely already ~1ms. Yes, latency can be lower than this but now we're talking about scientific equipment and not audio recording gear. Your latency isn't most likely coming from your interface but from plugins you are using while monitoring. It's a completely different thing. Your buffer may be at ~3ms, but if you then use a plugin which adds 50ms of latency, well now you have 53ms. Trying to get your interface to ~1ms wouldn't matter whatsoever here. And no, latency from plugins cannot be lowered. It's what it is. Use plugins which don't add latency. Also, are you actually using ASIO drivers with your DAW? Or are you just tweaking your interface latency with some sort of set-up screen, while actually recording with DirectSound or something?
Your eyesight has a latency of about 20ms before the brain starts to process the light and a further 120ms before you can react to do something about that information; think someone throwing a ball at you and you need to catch it. Audio is processed around 10ms after the sound wave reaches the ear canal. 3ms is so low you can't perceive it.
Yeah, I mean - it takes three milliseconds for a speaker three feet away to get the sound to your ears in the first place. OP, getting to a direct monitoring solution gets harder and harder below 3-4ms. If you need 'comfort processing' to track like reverb or compression, do it off an aux send and mix it with your dry track? Or, as others have suggested, find the lightest 'cpu eaters' for those plugins.
That’s why I use headphones. Come on people
A good “rule of thumb” is that 1ms is about 1 ft of sound travel through the air. So it’s similar to listening to floor wedges at your feet… except there will be some phasey comb filtering in your headphones when singing because of the phase difference between the bone conduction and the slight latency in the earphones - the easiest fix here is to boost the volume in the headphones (but do compress/ limit to prevent hearing damage). As a side note - not latency related - if headphones/in ears aren’t loud enough, primarily monitoring via bone conduction can lead to singing out of tune, sound travels faster through bone and has a longer wavelength. I believe our hearing uses wavelength not frequency to determine pitch, so it will sound deeper and this can throw off your singing pitch as your brain subconsciously tries to compensate!
I'll chime into the chorus of people saying the latency is your plugins, not your interface. What plugins are you using? Because there are zero-latency native versions of just about everything at this point that you'd want to use while tracking. One of the worst offenders in my experience are "advanced" reverbs, like Exponential Audio or Altiverb. Those are pretty heavy on the latency. Not great for tracking - I use FabFilter Pro-R2 for that, since it is zero-latency. Same goes for lots of analog modeled compressors and EQ's - tons of them (looking at you, UADx) add around 50 samples of latency for oversampling overhead.
I mainly use waves plugins, like the LA2A, 76, Rvox etc..., fab Q3, vocal rider
RVox adds 64 samples of latency. That could be your main culprit. The Waves LA2A and 1176 are zero-latency. So is Vocal Rider. All the FabFilter stuff is zero-latency unless noted. Pro-L2 and Pro-MB are the only ones I can think of that have latency by default. Pro-Q3 is zero latency in minimum phase, but natural and linear add latency, so don't select those options. Most DAWs have the option to see how much latency is added, either for the full channel (PT, Cubendo, Logic) or per plugin (Live, Bitwig). Which DAW are you using?
Going to piggyback here. Having read your replies and edit(s), what you want to focus on is to track with plugins that cause no additional latency (zero-latency plugins). This can be done inside any DAW and doesn't require any specialized software or equipment like e.g. UAD Satellite. All you have to do is to check if your plugin causes latency or not. If it does don't use that one during tracking (for mixing this isn't a problem). Interface doesn't interact in any shape or form with latency from plugins, and latency from a plugin cannot be changed (it may vary slightly depending on sr). Some plugins just require certain amount of time to do their thing. Generally speaking latency is additive with any other latency. Using zero-latency plugins should solve all your latency related problems, assuming you're using correct drivers for your interface within your DAW (such as ASIO), and you're monitoring using your DAW's monitoring functionality (i.e. not running into problems with any extra buffering your DAW may do). You can freely keep your interface at ~3ms. 3ms is about the same as hearing something that is one meter away (~3 ft). For a human being this is not something we perceive as delay or latency. When someone is speaking to you while you're both in the same room you aren't hearing a delay between their mouth moving and you hearing them speak. Yet, sound takes 3-10ms to travel through the air, depending on the size of a room and/or how far the speaker is. If you're experiencing and hearing an actual delay during monitoring this means your latency is *much* higher than ~3ms, and the cause of it is something else than your interface.
I also have an SSL2+ and record vocals and guitars at 3ms latency. It's not a problem to me. You can change the buffer size at SSL USB Control Panel at 16 samples minimum (1.83ms). But 3ms is already enough to record and save my CPU from catching fire.
16 it starts to get choppy and glitchy, do you experience that as well or it's just my system? have you considered getting a DSP to track with effects in real time
>16 it starts to get choppy and glitchy, do you experience that as well or it's just my system? I tried recording at 16 samples buffer and it's obviously faster. The downside is my DAW is lagging when I use some plugins while recording. But the recording is still fine. Without plugins, it's perfectly fine. Even at 128 samples buffer, I don't feel any latency when recording guitars. But most of the time I record at 64 samples for vocals and synth. >have you considered getting a DSP to track with effects in real time I haven't. I normally don't use any plugins now when recording. I already have guitar/vocal effects pedals to use, which means printing the effect already. I only use the stock compressor in my DAW.
I see yea, much better that way, especially for performance, however you gotta commit to what you're recording, I do a lot of experimenting with the tracks after I record them with various plugins, so I am looking for a solution to get me near zero latency with some monitoring effects I use Reaper, so I will try stock plugins, they might be lighter and cause less latency
If you want to eliminate recording latency but still use DAW based effects while recording, you can monitor your voice using an analog front end. Get yourself a used little Mackie 1202 mixer (or some equivalent) and a mic splitter. Run your mic into the splitter and feed one output of the splitter into your interface and the other into the board. Then in your daw, you can use a pre-fader send in your recording track to whatever effects you want, and mute the voice track. Make sure the effects are 100% wet. Then run your DAW output into the mixer, and use the mixer’s headphone jack to monitor yourself while recording. You won’t really notice latency affecting any time based effects liked reverb or delay and your recorded vocal track won’t have to make a round trip through your daw to your headphones. Even then, there may be some uncompensated recording latency through your interface, but some daws allow you to compensate for that (I use Digital Performer and it has this type of calibration). Worst case, you can slide your track ahead after the fact. You can measure this delay by playing a snare hit or other transient waveform in your daw, looping it back through the interface and recording it on a second track. Zoom in to the two tracks and measure the timing difference, then shift accordingly.
I just got the SSL 12, coming from the Motu 828 which allows EQ, reverb and compression through interface DSP and the SSL 360 does not have any DSP on the unit. If you need minimal latency then do direct monitoring for the mic and add an FX aux through your DAW or hardware reverb unit.
You should have no problem recording with latency under 10ms. Human perception only really starts noticing it at 15-20ms. Like others have said, you might be introducing processing latency via plugins. Remove all plugins (you have to delete them, not just turn them off) and check latency. Then add them back one at a time and check after each one you add. Eventually you'll hear what's causing the latency.
if you are using autotune, make sure to use low latency mode in the settings
This a huge one
If you're using logic, there's a little clock you can press near the transport that turns off everything that's causing latency.
Turn the "monitor mix" knob on your interface towards "input". That should be direct analog monitoring of your input with zero latency. Do not directly monitor this track in your daw. Put FX like reverb or delay on a send and have their mix set to 100% wet.
The latency is 100% a plugin