Latency

I am trying to record vocals in Logic, but I am getting some latency. I have my audio track routed to a bus with some reverb and delay. I turn on low latency mode but it bypasses the reverb and delay completely. If I turn off the actual reverb and delay plugin on the aux channel, you would think that the latency would go away but it doesn't. The only way I don't get latency is if I turn on low latency mode and don't have any reverb or delay. I have tried adjusting my buffer size and my recording delay is set to 0.

That's how it works.

You can set the bus to Low latency safe or insert a reverb on your track if you want.

 

You can also set Plugin Latency Compensation to Audio and Instrument tracks and then bypass the plug-ins that induce the most latency on your aux tracks.

Ok that makes sense. I guess that's why sends are sometimes orange. I have always wondered about that. How do I set plugin latency compensation?

Right click the send and select low latency mode (this is as well as running the global low latency mode)

if audio interfaces such as the UAD2 can do this now with no latency, with their native plugins, maybe it's time that finally, in 2016 this whole thing was redesigned to allow for truly latency free reverb plugins in software. I'm sure it can be done.

 

lol

Dealing latency is unfortunately one of the challenging issues of working with a DAW. It sounds like the reverb and delay plugin you are using are introducing quite a lot of latency. Different plugins require more latency then others, it depends on how much processing they are trying to do. When you have live audio you are feeding to an audio plugin, there is no way around having some amount of latency, but some plugins may be be less latent then others. However, its not arbitrary. The latency comes from the plugin doing its DSP number crunching to effect the sound in whatever cool way that it is.. It takes time for the computer to do that...and thus the latency.

 

So what can you do? Not much really. If the plugins you are using for your vocals are actually the ones causing the latency, which in this case it sounds like they are because when you turn them off, latency goes away. So it is what it is...but there are some working modes that you might find useful.

 

External monitoring. You feed your vocals WITHOUT fx directly back to you mixer/monitor before feeding into the sound card, so that basically you can hear your raw un-effected vocal sound without any latency, and without going through Logic's mixer. But set things up so that the vocals still go into Logic and still go through your reverb and delay...but you configure it so that Logic's mixer only outputs to your monitors the MIX part of the reverb (not the dry part). The end result is that your DRY vocals will have no latency and the reverb'd vocals will have a little bit of latency, but you may not mind it so much because its just the reverb.

 

If that still is too much latency to deal with, then try using external hardware reverb and delay just for when you are actually singing the vocal parts. Once they are recorded you can switch back to using plugins in your DAW and the latency will all be compensated for during playback without issue. Its just the live audio input that has the problem. You could also try some combination of the above using some cheapo reverb that doesn't have a lot of latency, just to get the part recorded, and then switch to the more elaborate reverb when you're playing back the tracks and mixing it.

Ok! Thanks for the help guys

Dealing latency is unfortunately one of the challenging issues of working with a DAW. It sounds like the reverb and delay plugin you are using are introducing quite a lot of latency. Different plugins require more latency then others, it depends on how much processing they are trying to do. When you have live audio you are feeding to an audio plugin, there is no way around having some amount of latency, but some plugins may be be less latent then others. However, its not arbitrary. The latency comes from the plugin doing its DSP number crunching to effect the sound in whatever cool way that it is.. It takes time for the computer to do that...and thus the latency.

 

So what can you do? Not much really. If the plugins you are using for your vocals are actually the ones causing the latency, which in this case it sounds like they are because when you turn them off, latency goes away. So it is what it is...but there are some working modes that you might find useful.

 

External monitoring. You feed your vocals WITHOUT fx directly back to you mixer/monitor before feeding into the sound card, so that basically you can hear your raw un-effected vocal sound without any latency, and without going through Logic's mixer. But set things up so that the vocals still go into Logic and still go through your reverb and delay...but you configure it so that Logic's mixer only outputs to your monitors the MIX part of the reverb (not the dry part). The end result is that your DRY vocals will have no latency and the reverb'd vocals will have a little bit of latency, but you may not mind it so much because its just the reverb.

 

If that still is too much latency to deal with, then try using external hardware reverb and delay just for when you are actually singing the vocal parts. Once they are recorded you can switch back to using plugins in your DAW and the latency will all be compensated for during playback without issue. Its just the live audio input that has the problem. You could also try some combination of the above using some cheapo reverb that doesn't have a lot of latency, just to get the part recorded, and then switch to the more elaborate reverb when you're playing back the tracks and mixing it.

 

They need to re-invent the entire system to eliminate or severely reduce latency to where it's not even noticeable. Of course it's possible on the UAD Apollo interfaces, but their plugins only run on their hardware. Surely an Intel CPU with 8+ cores can do it, there's nothing so special in that box that can't either be done natively. It's 2016, time for some ingenuity.. shouldn't have to use hardware reverbs to hear perfectly timed vocals.

 

lol

Surely an Intel CPU with 8+ cores can do it, there's nothing so special in that box that can't either be done nativelyl

 

That is not how it works.

UAD Apollo does not have to use the buffer that's needed when connecting to your DAW if you are monitoring through Console. This is the key difference.

There are latency issues and constrains inside the Apollo too.

 

• You can for example only use a maximum of 4 plug-ins on a channel strip depending on which plug-ins you insert because each channel is limited to use one single chip.
   
   
  
• You will have additional latency added by plug-ins that are up-sampled. Here's the small list:
  
  • 
     
    
  • AKG BX 20 Spring Reverb
    
  • Ampex ATR-102
    
  • Bermuda Triangle
    
  • Ibanez Tube Screamer TS808
    
  • Raw Distortion
    
  • API 550A
    
  • API VisionChannel Strip
    
  • Fairchild Tube Limiter Collection FATSO Jr./Sr.
    
  • Manley Massive Passive
    
  • Manley Variable Mu
    
  • Moog Filter
    
  • Neve 1073 (newer model w/preamp) Neve 33609
    
  • Neve 88RS (newer model w/preamp) Oxide Tape Recorder
    
  • Pultec EQP-1A
    
  • Pultec MEQ-5
    
  • Studer A800
    
  • Thermionic Culture Vulture
    
  • UA 1176 Limiter Collection
    
  • UA 610 Tube Preamp & EQ Collection
    
  • Precision Limiter
    
  • Precision Maximizer
    
  • API 560
    
  • Teletronix LA-2A Leveler Collection
    
  • Helios Type 69
    
  • Harrison 32C
    
  • Neve 1073 Legacy, 1081, 31102 Precision EQ
    
  • Pultiec EQP-1A Legacy Pultec-Pro Legacy
    
  • Pultec HLF-3C
    
  • SSL E Channel Strip
    
  • Trident A-Range
    
  • Precision Multiband
    
  • Little Labs IBP
    
  • Lexicon 224
    
  • EMT 250
    
  • MXR Flanger/Doubler

   

  [*]Using IDC will induce latency on all channels similarly to Logic's PLC All preference.

   

   

  [*]Adding an aux will add additional latency

 

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You can use a software reverb in Logic while monitoring via your interface. The reverberated signal will have a some latency on it but that will often act in our favour as the pre-delay.

 

They need to re-invent the entire system to eliminate or severely reduce latency to where it's not even noticeable. Of course it's possible on the UAD Apollo interfaces, but their plugins only run on their hardware. Surely an Intel CPU with 8+ cores can do it, there's nothing so special in that box that can't either be done natively. It's 2016, time for some ingenuity.. shouldn't have to use hardware reverbs to hear perfectly timed vocals.

 

lol

 

That's a nice dream, but if they could have done that they would have done it a long time ago. The reason LPX and other DAW's all have to use an audio buffer, and why some plugins need to impose even more latency on top of that, is because your computer is doing hundreds of other tasks at the same time and its time sharing the CPU and other data busses between LPX, MacOS, plugins running inside LPX, hardware drivers, screen updates, your mouse, etc.. All these different pieces of software are taking turns using the CPU and data busses of your computer. Because of that, there is no software on a mac that can produce uninterrupted real time audio streams in absolute zero latency real time. Its simply not possible. Its not only a matter of how powerful the CPU is, its also a matter of priority. When you have a multi-tasking operating system and DAW, then you can either have audio with a lot of drop outs and glitches, or you can use a buffer.

 

An audio buffer is used so that the DAW can work on one buffer full of audio at a time. It gets its chance to use the CPU, and fills up the buffer. It might have to use more than one trip back and forth to the CPU, taking turns with other processes on your mac, until finally it fills up the buffer. As long as the sound card always has some data in the buffer, it can keep sending un-interrupted drop-free audio to you speakers. If the software processing is too slow to fill the buffer to keep up with how fast the sound card is sending it to your speakers and the buffer dries up, then that is when you get audio drop outs.

 

The only way you could avoid using an audio buffer would be with extremely dedicated hardware that doesn't have to time share the CPU and other hardware resources with other processes, like all programs running on a mac do. So external gear is very dedicated with a single CPU path. Not multi-tasked. There is still some latency there even in those, but they are so optimized and prioritized, that that they can use a very tiny sized audio buffer, the processing doesn't have to be timeshared per say..

 

On the mac, it has to be time shared and there will always have to be a buffer. To me the fact that some people are able to get 32 sample latency, or in some cases 16 sample latency without audio dropouts is simply amazing, That is some pretty serious CPU crunching and data throughput in the Mac's busses to achieve things fast enough that within 16 or 32 samples, a host like LPX can do everything it needs to do and keep the buffer filled, even while everything else that is happening at the same time on the computer and stealing time share time.

You can use a software reverb in Logic while monitoring via your interface. The reverberated signal will have a some latency on it but that will often act in our favour as the pre-delay.

 

This ^^^^ +1

That ^^^^

 

Or split the signal and add reverb outside the box. Singers will love you, promise. My wife does

That is not how it works.

UAD Apollo does not have to use the buffer that's needed when connecting to your DAW if you are monitoring through Console. This is the key difference.

There are latency issues and constrains inside the Apollo too.

 

I do know how it works, I'm saying with some ingenuity, it can be done, Apple would have to do it of course.

 

I'll combine your posts for the interest of brevity:

 

That's a nice dream, but if they could have done that they would have done it a long time ago. The reason LPX and other DAW's all have to use an audio buffer, and why some plugins need to impose even more latency on top of that, is because your computer is doing hundreds of other tasks at the same time

 

The main reason they couldn't do it a long time ago was that CPUs weren't very fast, now they are, and the whole thing needs to be redesigned. Also:

 

The only way you could avoid using an audio buffer would be with extremely dedicated hardware that doesn't have to time share the CPU and other hardware resources with other processes

 

Compared to decades ago, those other tasks are negligible in the world of 12x 3 GHz CPU cores. But even if they were, audio is important enough to dedicate cores to. Nothing else going on in the background, just processing audio. Another core drawing the screen, etc. Not just split it all out shared amongst spotlight indexing the hard drive and getting the latest weather and stocks and sports and that other crap and hope for the best.

 

I'm talking about a real dedicated OS X + Logic SINGLE TASKING computer! That doesn't buffer the same way and with a dedicated chip inside of it like the Apollo's, if necessary.

 

We had "workstations" in the 90s that did nothing else, so why can't computers finally do this again so that latency is practically non-existant?

 

It'll never happen, because no one thinks about this stuff. They just throw in workarounds like an orange "low latency" button and call it a day.

 

lol

its not just about the CPU's needing to be fast, its also about the fact they are multi-tasking operating systems. its not gonna happen as you wish

its not just about the CPU's needing to be fast, its also about the fact they are multi-tasking operating systems. its not gonna happen as you wish

 

Like I said, this is the wrong approach. They need to either not multi-task so much anymore or provide huge priorities to the foreground application.

 

It would probably even require the Logic team to takeover Apple. haha

 

lol

you would not have a useable operating system at all if you tried to make OSX non-multi tasking. Its the very essence of what a modern OS needs to be and it is this way at the core. If you want a single process, then you don't want an operating system, you want a dedicated device... OSX is based on BSD unix and is a multi-tasking operating system, end of story. This has been found to be the best way to approach computing in general. In the past win95, WinNT, DOS...those were less multi-tasking...with numerous problems. When we make the jump to Windows2000 and UNIX based systems, everything became multi-tasking at the core. Attempts to get around it create more problems then they solve. Its better to just accept that fact that we are going to have audio buffers and latency probably for the next few decades at least in terms of audio production on a Mac or PC computer. If you want something else you'd need to write a different OS that is basically more like LPX running AS the OS itself and nothing else, essentially turning the mac into a dedicated hardware device. Then maybe it would be possible to do as you are wishing.

you would not have a useable operating system at all if you tried to make OSX non-multi tasking. Its the very essence of what a modern OS needs to be and it is this way at the core. If you want a single process, then you don't want an operating system, you want a dedicated device... OSX is based on BSD unix and is a multi-tasking operating system, end of story. This has been found to be the best way to approach computing in general. In the past win95, WinNT, DOS...those were less multi-tasking...with numerous problems. When we make the jump to Windows2000 and UNIX based systems, everything became multi-tasking at the core. Attempts to get around it create more problems then they solve. Its better to just accept that fact that we are going to have audio buffers and latency probably for the next few decades at least in terms of audio production on a Mac or PC computer. If you want something else you'd need to write a different OS that is basically more like LPX running AS the OS itself and nothing else, essentially turning the mac into a dedicated hardware device. Then maybe it would be possible to do as you are wishing.

 

I don't think you know what you're talking about..

 

Windows NT was the first PRE-EMPTIVE multi-tasking OS in the Windows world, it was also multi-threaded. That's where the big jump was, not in Windows 2000.

 

DOS did not multi-task, at all. Windows 3.x, in the 16-bit era was COOPERATIVE multi-tasking, there's a big difference.

 

But, regardless, this was all in an era when CPU resources were scarce, so all that buffering had to be done. We're not in that era anymore, but the way DAWs function hasn't changed much. They should be radically redesigned.

 

Computing hardware is such a commodity that it's not uncommon for people to own multiple PCs and devices anyway, why can't we go back to single tasking units? For those who can afford it.

 

lol

ok thanks for letting me know that I don't know what I'm talking about. but my friend, what you are desiring is not possible today on modern OS. Sorry.

ok thanks for letting me know that I don't know what I'm talking about. but my friend, what you are desiring is not possible today on modern OS. Sorry.

 

You seem knowledgable, but I was correcting some details in your statements. It is indeed not possible, that's why I want to see a re-design.

 

Is it even possible to process audio in a computer with no buffers whatsoever? That's where I would begin. Imagine it, audio is always guaranteed on-time with never any popping, drop-outs or under-runs.

 

This would be done either through dedicated cores or dedicated DSPs inside the machine. But no one cares enough about the problem to make it happen.

 

lol

Hey All,

 

I have been having a persistent vocal recording latency that trumps everything mentioned above, as I seem to get the same latency even when recording a totally "dry" vocal with all of the latency controls adjusted to the lowest settings and Low Latency Mode switched on.

 

I recently did a clean, manual re-install, so I am pretty sure this isn't a larger problem.

 

Is there anything I'm missing?

 

I'm using the most up-to-date version . . .

 

Thanks!