Audio delay with interface

[Brawders]

I am by no means a Pro Tools expert, just a humble guy trying to record things in my spare room. I could be totally wrong in the way I'm understanding the flow of this technology and I will admit that. Let me give you a run down of all of the plug-ins I have and how I have my interface plugged in.

Mac 3.1 running Mojave 10.14.6

ProTools 2020.5.0

Focusrite Scarlett 18i8 optical to a Focusrite Scarlett Octopre. 18i8 is hooked up to the Mac via a USB 2.0. My two studio monitors are also hooked into the outputs of the 18i8.

Plug-ins in my session right now:

EQ3 7-Band
Pro Compressor
Pro Subharmonic
Dyn 3 Compressor/Limiter
Studio Reverb
Bomb Factory 76
D-Verb
Fuzz-Wah
Studio Reverb
Dyn 3 De-Esser
Eleven MK II
POW-r Dither

(All included with the version of Pro Tools I got)

Ozone 9 Elements
Vocal Doubler

(Found free online)

Playback Engine: Scarlett 18i8 USB
H/W Buffer Size 64 Samples
Dynamic Plug-In Processing on
Cache Size Normal

I am plugging my guitar straight into the front 1/4" on the 18i8. I am using the Eleven as my plug-in.

I have been muting the channel on my mixer and playing along with the recording, the only problem is when I go to listen to it back the guitar I just recorded is a second or so late and I have to fudge it around to line it up with the rest of the tracks. If I plug some headphones into the 18i8 I still get the delay when listening to my guitar through them.

Anything else I can provide to make me less of an idiot when it comes to all of this?

[Darryl Ramm]

Again the most important info is just buffer size and sample rate. You are using a 64 samples IO buffer but at what sample rate?

And you divide them and get the expected latency... which is... WHAT?

and add in some conversion latency (you can probably look up the specs of your interface or find that on the web) and that is what you should see. And it should be a small latency of a few-handful of ms, again equivalent to a typical guitar amp cab not too far from a performer. So while you are playing... are you having problems monitoring/playing along with the pre-recorded content at 64 sample IO buffer?

Now you are talking about having to move things in Pro Tools to make things line up. Again it's an audio workstation.. full of things like time rulers... so how far exactly in ms are you having to move samples to get things to line up. Pro Tools standard does not time shift recordings to correct for the converter latency (Pro Tools Ultimate with Avid hardware will correct for more there). So there is always some small offset. But you were talking about "seconds". So what is it exactly? What do you measure in Pro Tools?(set the time display in milliseconds, drag the "I" cursor between two events and the time display up the top of the mix window will tell you that time difference).

You also did not clarify what version of Pro Tools 7 you have been using before. If you have only used Pro Tools 7HD that could explain your confusion.

Originally Posted by Brawders

/snip/
Plug-ins in my session right now:

EQ3 7-Band
Pro Compressor
Pro Subharmonic
Dyn 3 Compressor/Limiter
Studio Reverb
Bomb Factory 76
D-Verb
Fuzz-Wah
Studio Reverb
Dyn 3 De-Esser
Eleven MK II
POW-r Dither

OK but get yourself sorted out and understand what is going on on a one track session with only the amp sim plugin. As you ad more complex routing and plugins you risk adding plugin latency. Get things working and understood with a one track session before trying to do more.

In addition to the IO buffer latency, and convertor hardware latency some of the plugins will have latency, occasionally that can be quite high, but there is no horrific plugin here. But here is a handy respource anyhow https://docs.google.com/spreadsheets...hC0/edit#gid=0

And when you start dealing with plugins you often need to turn on delay compensation, all that does is make all the signal path latencies equal to the one with highest latency, it does not correct IO buffer or converter latency. And as you play with plugins and delay compensation you will be able to see the delay values reported in the delay compensation values in the mixer window.

And again you can do more wrong here than just use high latency plugins, you can be screwing things up trying to use aux tracks and suffer an additional 1024 or 2048 sample delay for the fixed playback buffer. We can't guess what you are doing, and neither can we guess what actual latency you have because you've not ever mentioned an actual measured accurate number.

Ozone 9 Elements
Vocal Doubler

(Found free online)

Playback Engine: Scarlett 18i8 USB
H/W Buffer Size 64 Samples

Dynamic Plug-In Processing on <- makes no difference to latency
Cache Size Normal <- disk cache, makes no difference to latency (but why have this set to "normal" on a machine with so much memory... it should be set large enough to fully cache your sessions?)

I am plugging my guitar straight into the front 1/4" on the 18i8. I am using the Eleven as my plug-in.

I have been muting the channel on my mixer and playing along with the recording, the only problem is when I go to listen to it back the guitar I just recorded is a second or so late and I have to fudge it around to line it up with the rest of the tracks. If I plug some headphones into the 18i8 I still get the delay when listening to my guitar through them.

You mean muting in the Focusrite Mix Control software... that's great, and correct thing to do if you want to "monitor wet" (ie. through an amp sim).

Again with the "second or so"... are you actually measuring that?

It's next to impossible to follow you. What session are you talking about here? The large one... forget that for now. On a small session with just one track and one Eleven plugin what exact latency do you measure in Pro Tools? How much in ms, measured in Pro Tools, are you having to move things to align?

And on top of the conversion latency, and IO buffer latency, you may have some performer latency, but nothing here should be near a *second*.

[Brawders]

Originally Posted by Darryl Ramm

Again the most important info is just buffer size and sample rate. You are using a 64 samples IO buffer but at what sample rate?

And you divide them and get the expected latency... which is... WHAT?

and add in some conversion latency (you can probably look up the specs of your interface or find that on the web) and that is what you should see. And it should be a small latency of a few-handful of ms, again equivalent to a typical guitar amp cab not too far from a performer. So while you are playing... are you having problems monitoring/playing along with the pre-recorded content at 64 sample IO buffer?

Now you are talking about having to move things in Pro Tools to make things line up. Again it's an audio workstation.. full of things like time rulers... so how far exactly in ms are you having to move samples to get things to line up. Pro Tools standard does not time shift recordings to correct for the converter latency (Pro Tools Ultimate with Avid hardware will correct for more there). So there is always some small offset. But you were talking about "seconds". So what is it exactly? What do you measure in Pro Tools?(set the time display in milliseconds, drag the "I" cursor between two events and the time display up the top of the mix window will tell you that time difference).

You also did not clarify what version of Pro Tools 7 you have been using before. If you have only used Pro Tools 7HD that could explain your confusion.



OK but get yourself sorted out and understand what is going on on a one track session with only the amp sim plugin. As you ad more complex routing and plugins you risk adding plugin latency. Get things working and understood with a one track session before trying to do more.

In addition to the IO buffer latency, and convertor hardware latency some of the plugins will have latency, occasionally that can be quite high, but there is no horrific plugin here. But here is a handy respource anyhow https://docs.google.com/spreadsheets...hC0/edit#gid=0

And when you start dealing with plugins you often need to turn on delay compensation, all that does is make all the signal path latencies equal to the one with highest latency, it does not correct IO buffer or converter latency. And as you play with plugins and delay compensation you will be able to see the delay values reported in the delay compensation values in the mixer window.

And again you can do more wrong here than just use high latency plugins, you can be screwing things up trying to use aux tracks and suffer an additional 1024 or 2048 sample delay for the fixed playback buffer. We can't guess what you are doing, and neither can we guess what actual latency you have because you've not ever mentioned an actual measured accurate number.



You mean muting in the Focusrite Mix Control software... that's great, and correct thing to do if you want to "monitor wet" (ie. through an amp sim).

Again with the "second or so"... are you actually measuring that?

It's next to impossible to follow you. What session are you talking about here? The large one... forget that for now. On a small session with just one track and one Eleven plugin what exact latency do you measure in Pro Tools? How much in ms, measured in Pro Tools, are you having to move things to align?

And on top of the conversion latency, and IO buffer latency, you may have some performer latency, but nothing here should be near a *second*.

Ok I have some more answers.

Sample rate: 44.1
Buffer size: 128 (if I try 64 I get AAE -6101)

Yes while I'm playing with it the way it's set above there's a delay and it's very hard to play along with.

Original recording I did started at 0:05.723 and I had to shimmy it over to 0:05.578 for a difference of 0.152. Noticeable to me and annoying.

I was working with Pro Tools 7 LE before, and my biggest project was like 12 tracks, hardly any plug-ins.

I do have a lot of my tracks routed to different places. For example I have all of my guitar tracks on two busses, one for a guitar bus, and the other for a guitar parallel compression. Then I have those two tracks bussed to another aux input to just control the entire fader for the guitars. So I can mix the guitars together how I want, then control their overall volume from one place.

When I say it's off by a "second" it's probably a lot less than that, but enough to throw me off when I'm trying to play along with it.

When I say I mute the track it's not in the Focusrite mix control software, it's in my channel strip in my ProTools session.

I opened up a brand new session with the same parameters: 44.1 and 128. I recorded one track by itself and then another, and they pretty much lined up perfect. I didn't really have to adjust anything.

So if I'm understanding this correctly, and I may not be. It could be a combination of having too many plug-ins and routing into Aux tracks that could be slowing me down? If I'm recording into these bigger tracks then do I just need to turn on delay compensation and low latency monitoring? I found those two when I was looking up something you said.

Is there any other info I can give you?

[Darryl Ramm]

Originally Posted by Brawders

Ok I have some more answers.

Sample rate: 44.1
Buffer size: 128 (if I try 64 I get AAE -6101)

Yes while I'm playing with it the way it's set above there's a delay and it's very hard to play along with.

Original recording I did started at 0:05.723 and I had to shimmy it over to 0:05.578 for a difference of 0.152. Noticeable to me and annoying.

I was working with Pro Tools 7 LE before, and my biggest project was like 12 tracks, hardly any plug-ins.

I do have a lot of my tracks routed to different places. For example I have all of my guitar tracks on two busses, one for a guitar bus, and the other for a guitar parallel compression. Then I have those two tracks bussed to another aux input to just control the entire fader for the guitars. So I can mix the guitars together how I want, then control their overall volume from one place.

When I say it's off by a "second" it's probably a lot less than that, but enough to throw me off when I'm trying to play along with it.

When I say I mute the track it's not in the Focusrite mix control software, it's in my channel strip in my ProTools session.

I opened up a brand new session with the same parameters: 44.1 and 128. I recorded one track by itself and then another, and they pretty much lined up perfect. I didn't really have to adjust anything.

So if I'm understanding this correctly, and I may not be. It could be a combination of having too many plug-ins and routing into Aux tracks that could be slowing me down? If I'm recording into these bigger tracks then do I just need to turn on delay compensation and low latency monitoring? I found those two when I was looking up something you said.

Is there any other info I can give you?

152ms @ 44.1k is .152 * 44,100 = 6700 samples.

That's a *lot* and none of the plugins you have listed should get you there. A single aux misuse will get you 1024 samples latency at 44.1kHz (Aux use separate buffer size of 1K, 2K or 4K depending on sample rate, you have no control over that), maybe you may have managed to route multiple of those together, who knows. But assuming your number is correct that's so large it's going to be something you are doing very wrong.

But now you know, nothing here was "a second" or "two". Nothing here was "Audio delay with interface", at small IO buffers simple sessions work seems to work as expected... all this time wasted.

And you have more complex sessions that are doing something, what exactly I have no idea what is causing problems. Start over, follow some tutorials if you need to. Keep things as simple as dirt until you learn more, don't use aux tracks on signal paths you are recording, e.g. understand this thread and the linked talk http://duc.avid.com/showthread.php?t=392221.

[albee1952]

Some general info to add to the discussion:

1-Ozone 9 is a serious resource hog, and has a ton of latency, so best to leave it inactive until mixdown(when you can up the buffer to 1024). Remember that "inactive" is not the same as in Bypass

2-Plugin latency on a master track is NOT compensated for, so be thoughtful on what plugins you have on any master track while recording. I make sure that Delay Compensation shows no more than 10-11 samples while recording(any plugins that add more latency stay inactive until its time to mix).

3-There is at least 1 way to get bogus latency with ADC and that is to have a send that goes nowhere. Example: You put a send on a guitar track to feed an aux track with a delay. Then you change your mind and delete the aux track, but leave the send. That will cause a bunch of bogus ADC if you try to record.

4-Which version of PT are you using? Is it compatible with a Mac Pro 3,1? Wondering if this is a case of an old computer trying to run new software. IIRC, the 3,1 uses slow RAM(like 667?), and you said you have 20GB. That means mis-matched RAM sticks, which often hurts performance. You also elude to some kind of patch to make Mojave work? That could add to the problems. It might be time for an upgrade on the old Mac

5-More info about latency: The buffer setting is how much time you allow the system to process audio and send it out. Higher buffer(like 1024)=higher latency. Lower buffer(like 64)= lower latency. This amount is added to the latency from A>D and D>A(which is a fixed amount, depending on the sample rate). If your computer cannot record at the 64 or 128 buffer setting, then your computer is not up to the task. Since its almost a 13 year old computer, it might be much happier on a much older OS and version of Pro Tools(that could mean some plugins would not work, depending on how far back you need to go with software and OS).

6-As said, the specs on your drives matter, but this is not about latency, its about performance(which includes that pesky buffer setting). If you have a single SATA drive and you have everything on it(OS, Pro Tools, sessions), then its not going to perform well at all(maybe okay at 1024 buffer, but you need to be able to use the 64 buffer for recording). A 7200 rpm drive will not do very well. A 5400 rpm drive will fall on its face completely. If you only have a single drive, then you really need to add at least 1 more fast drive for sessions(7200 rpm or SSD on an adapter sled). If you have any big sample libraries, they would be best if they were on a 3rd separate drive(also 7200 rpm, or SSD). Given the age of the Mac, will a 3,1 net any benefit by using SSD's? That's something else to consider.

[originalscottyg]

Originally Posted by albee1952

Some general info to add to the discussion:

1-Ozone 9 is a serious resource hog, and has a ton of latency, so best to leave it inactive until mixdown(when you can up the buffer to 1024). Remember that "inactive" is not the same as in Bypass

This strikes me as the most likely culprit. High latency plugins on a master or in the monitor path.

[Brawders]

Lemme try to answer a few of the questions, yes it’s a 3.1 with a patched version of Mojave. Basically just a way to run Mojave on a little older computer.

The RAM I have is the 800 MHz. 4 x 4 gb, and 2 x 2 gb. Memory is good as long as it’s in pairs, and not mismatched.

From what the last few people have said I think it is the ozone that’s bogging me down. I have it on quite a few of my tracks to sweeten them up a little. That plus me trying to record on tracks that are routed to busses I think makes for the problem.

I will have to go back and make the ozone plug in inactive and record straight to the main outputs and I think that should fix it.

[originalscottyg]

As mentioned above, make sure that delay compensation is turned on.
Go to the menu Options->Delay Compensation and make sure it is checked.

Then go to the menu View->Mix Window Views->Delay Compensation and make sure that is checked as well.

That will show a small delay compensation window below the faders in the Mix Window.

Numbers that are ORANGE indicate the channel or channels with the most delay in the session and numbers that are RED indicate that you have exceeded the system's maximum amount of delay compensation.

[Brawders]

Thank you everyone for the answers and help. I feel a little smarter now.

[djanogil]

Did you sort out your problem in the end?
I have been having the same issue since upgrading to PT2020 on an older Mac Pro 5.1.