Early audio with destructive record (delay compensation incorrect?)

[Matthew Steel]

A recent project was a live performance where I recorded individual tracks, stems, and a live mix.

A few spots needed fixes so I set up the session to record to a mixdown track. Initially the mixdown track contained a file created by consolidating the live mix to get a separate audio file. Then I used destructive record to print changes into the mixdown track.

I found that the places where I made fixes, the new audio was early by some number of samples. The biggest problem with that is that it creates discontinuities at the in and out points and often these are audible as clicks.

My initial setup was a few tracks with Trim plugins, and an outboard reverb with a send to a hardware output and returning on an aux in track. Delay compensation was on, and the aux input showed a compensation value of 512; with this setup the printed audio was always 587 samples early.

I tried lots of variations of DSP/Native modes for various tracks, inserting dummy trim plugins like I used to do in PTLE for manual delay compensation, disabling plugin compensation for the aux track. Some things changed the number of samples but with delay compensation the audio was always early.
Without delay compensation things were late, which was to be expected but is also not useful.

The only way I could get the mixdown track to sync was to disable the aux input. Then the alignment was perfect. Of course that was not useful if I actually needed the reverb for a particular fix.

At one point I thought I had solved the issue by powering down the Mac Studio (and the chassis), but later that definitely did not fix the problem.

One thing I did not try (yet) was a regular non-destructive punch to see if this is a more widespread issue.

Any tips? I suppose I could slide the mixdown track the requisite number of samples and then the new audio might line up with the old...

[this is on Ultimate 2022.10 under Rosetta on a Mac Studio M1 running Monterrey with an HDX card in an Avid TB->PCI chassis]

[its2loud]

Originally Posted by Matthew Steel

A recent project was a live performance where I recorded individual tracks, stems, and a live mix.

A few spots needed fixes so I set up the session to record to a mixdown track. Initially the mixdown track contained a file created by consolidating the live mix to get a separate audio file. Then I used destructive record to print changes into the mixdown track.

I found that the places where I made fixes, the new audio was early by some number of samples. The biggest problem with that is that it creates discontinuities at the in and out points and often these are audible as clicks.

My initial setup was a few tracks with Trim plugins, and an outboard reverb with a send to a hardware output and returning on an aux in track. Delay compensation was on, and the aux input showed a compensation value of 512; with this setup the printed audio was always 587 samples early.

I tried lots of variations of DSP/Native modes for various tracks, inserting dummy trim plugins like I used to do in PTLE for manual delay compensation, disabling plugin compensation for the aux track. Some things changed the number of samples but with delay compensation the audio was always early.
Without delay compensation things were late, which was to be expected but is also not useful.

The only way I could get the mixdown track to sync was to disable the aux input. Then the alignment was perfect. Of course that was not useful if I actually needed the reverb for a particular fix.

At one point I thought I had solved the issue by powering down the Mac Studio (and the chassis), but later that definitely did not fix the problem.

One thing I did not try (yet) was a regular non-destructive punch to see if this is a more widespread issue.

Any tips? I suppose I could slide the mixdown track the requisite number of samples and then the new audio might line up with the old...

[this is on Ultimate 2022.10 under Rosetta on a Mac Studio M1 running Monterrey with an HDX card in an Avid TB->PCI chassis]

Any time you use hardware inserts you’re going to have uncompensated latency. This has to be manually configured and calculated. Automatic Delay compensation only works for “in the box” latency.

You can read more here

https://duc.avid.com/showthread.php?t=418001

And here

https://avid.secure.force.com/pkb/ar...e-Insert-Delay

[Darryl Ramm]

Lots wrong here.

What exact make/model is this outboard reverb? Presumably it is digital and comes with its own latency... which is what? from its specs/documentation.

My initial setup was a few tracks with Trim plugins, and an outboard reverb with a send to a hardware output and returning on an aux in track.

Why are you trying to do this and not use a Pro Tools Hardware Insert like God intended? All described clearly in the Pro Tools reference guide.

The interface IO latency of a HW Insert *is* compensated for when you enable delay compensation (it is an insert, delay compensation compensates for software and hardware inserts) but not any outboard latency in the HW insert... Pro Tools cannot guess what that is and unlike many modern DAWs does not have a ping function to measure that latency, so as pointed out you need to configure that manually.

Any negative time offset might be caused by known delay compensation bugs in Pro Tools, on pure native systems disabling low latency monitoring or disabling ignore errors avoids some known bug(s) (including with sends), on HDX based systems I have no idea. I have documented those bugs on DUC and wasted my time filing bug reports for some of them. How Avid can have so many issues with delay comp and not fix the bugs is beyond me. But you should at least start by getting this setup and done properly with a HW insert or even better... just give up and use a software plugin....

All this faffing around to go outboard to just get to a reverb seems a waste of time when you have no end of wonderful reverbs available in plugins all that are much easier to use, come with sample accurate delay compensation, do not have known delay comp bugs, etc. Now if you are sending stuff through an actual vintage EMT plate reverb, or are sitting in Capitol studio with access to their echo chambers... I'll be impressed, but still nobody will know any of that is not a plugin.

[Matthew Steel]

OK, I understand what you're saying about setting up the reverb as a hardware insert, but in this case it's not so simple. The reverb is a Lexicon 480L. It has two inputs and 4 outputs. It can be operated in stereo->stereo or dual mono->stereo modes.

Unless things have changed, Pro Tools requires hardware inserts to have the same width on input as output, and to use the same channel numbers for input and output on the interface. These limitations are why it is set up as send/return. I really don't know a better way for this particular piece of equipment.

Plus, it's a reverb. I'm not expecting Pro Tools to line that up. At least within a certain range uncompensated delay just translates to a longer predelay and I can easily live with that. Pro Tools doesn't know what is coming into that aux in, it could just as easily be me singing karaoke.

What I am expecting Pro Tools to do is to line up the dry elements that never leave the Pro tools mix engine. The delay compensation engine is correctly adjusting for the plugins and keeping things in phase. It is even trying to compensate for the total processing latency. The reason I know that is because the resulting audio is early in the timeline.

[Darryl Ramm]

Again my first choice would be to go get a reverb plugin and move on. Unfortunately if you exactly want a 480L UAD's 480L clone is UAD2 DSP only. And Lexigone's own plugins (they *don't* have an overall 480L clone, but they have stand alone algorithms) look like abandonware.

You *can* insert a mono->stereo HW insert. You put the HW inset on a stereo track with the mono input signal recorded to/panned to the one track/output you are feeding into the outboard box input. Yes that means you burn one output port you are not using. The benefit of doing things this way is a consistent workflow with inserts and you have H/W insert delay compensation that (if things are working OK, which they may not be) lets you correct for outboard latencies. It also stops folks on DUC complaining you should use a HW insert

You can spend time trying to debug what is going on on your system. I've been there and done that for native systems. There are multiple known bugs in delay comp related issues. And those sure can result in an apparent negative offset. I've discussed this in the past in probably tens of posts on DUC. And nothing you are describing here excludes the type of problems already known, including the negative offset/"early in the timeline" behavior... caused by the delay compensation engine overcompensating for the path the "insert" signal is taking.

Some testing suggestions: Get HDX out of the picture, do you have a native interface you can test with? Set ignore errors and low latency monitoring off and see what happens. Clone the session and reduce it as much as possible to one or two tracks and deleting (not just hiding) everything else. Try with and without "delay" compensation enabled.

[Darryl Ramm]

Oh and also try unchecking

Setup>Pro Tools Preferences>Mixing>Allow Sends to persist during LLM

The introduction of that feature/option seemed to create a latency compensation bug, at least on native systems.

[Matthew Steel]

(I don't think the HDX playback engine has Low Latency Monitoring in the same way as native systems do. In any case that entry isn't in my Options menu and the checkbox you mentioned isn't in my Preferences)

I did some more troubleshooting by making a simple session from scratch:

* One mono audio track (Audio 1) with one test clip for playback, fader at unity, assigned to Bus 1.
* One mono aux input (Aux 1) from an interface, muted, assigned to Bus 1.
* One mono audio track (Audio2) with input from Bus1.
* No plugins or sends anywhere in the session.
* Record armed Audio 2 (normal record mode).
* Selected a section to record.
* Hit F12 to record.

Results should be that the newly recorded clip on Audio2 is identical to the test clip.
Instead, the audio on Audio2 is early by 587 samples. This value seems closely related to the total system latency.

Before testing with a non HDX system as suggested, I decided to try HDX with the classic (non-hybrid) engine.

The results of the test were perfect alignment of the file.

That leads me to believe that the issue is related to the Hybrid engine. I do not have a different HDX system available (to see if it is just my system) or a Carbon system (to see if it is just HDX Hybrid).

From my testing, conditions required to experience this behavior are:
1) Have a track with input from a hardware interface. An aux track and an audio track in input mode both behaved the same for me. A hardware insert does not appear to cause the behavior.
2) The track must be in the mix path.
3) You must be using the Hybrid Engine (I do not know if HDX or Carbon makes a difference).
4) You must be recording to a track. Bouncing online and importing the track shows perfect alignment. Offline bounce also shows perfect alignment, but of course offline bounce with a hardware input that actually contributes to the mix seems pointless.

So this is probably a bug. But I have possible 3 workarounds:
1) Plugins for reverb
2) Hardware insert
3) Don't use Hybrid Engine