Buggy Automatic Delay Compensation

[brunogermain]

Hello everyone,
first post on the forum.
I've been using Pro Tools for almost 10 years now, and only yesterday I've had my first serious issue.

I run Pro Tools Ultimate 2019.12 with Mojave 10.14.6 on a MacPro5,1 with updated firmware and GPU (Radeon HD7950), audio interface is a MOTU PCI-424 with 3x Motu HD192 converters.

I always work at 96kHz 24bit, tracking on both tape and directly to digital, depending on the project.

In the last couple of years I switched from mixing on the desk ITB, using Hardware Inserts (with sample accurate delay compensation) even for parallel compression. Never had any kind of issue with this workflow. Never experienced any kind of phasing issue, if the HW Insert was properly configured according to the sample rate.

Until yesterday...

A client sent some tracks to mix; the usual post rock session: Drums (10 mics), bass (2 tracks), 4 guitar tracks and a couple of synthesisers.
All was recorded @ 48kHz 24bit using Cubase. I just got the WAV's and imported them in newly created PT session.

To mix the drums, I made 4 busses (Kick, Snare, Overheads and a Parallel compression): each bus had a HW Insert on it.
Kick, Snare and OH were going directly to the master and sent (with a pre fader send) to the Parallel Compression Bus and to Reverb, which both went to the master. Pretty standard routing, everything sounding as it should.

The master goes (physically) to an SSL Bus Compressor and back into a Stereo track always set to Input Monitor, which will record my final mix.

At some point, while playing back, the ADC starts literally drifting, as if Delay Compensation was manually being turned on and off, randomly. It would play fine for a couple of minutes, and then you can definitely hear the snare sound become muffled and then coming back to normal.

Luckily the client wasn't there, so I had time to try some empirical workarounds.
I started disabling Plugins, trashing Pro Tools Preferences, disabling and enabling Dynamic Plugin Processing, fiddle with Disk Cache, different playback buffer settings, all to no avail. The CPU load never exceeds 10%.

So I thought that the only difference from my usual workflow was the session's sample rate, and never had such an issue working at 96k, even with much heavier sessions.

I created a blank 96k session and hit Import Session Data from the 48k session, converting all the audio files.

All well, Delay Compensation is steady and never drifts.

Anyone might know why?

[albee1952]

Some things to consider:
1-have you manually entered a delay time for your hardware insert(s)? This is not automatic in Pro Tools(and may be different based on the sample rate used).
2-the different sample rate might be "the" difference as the lower rate will double the latency as compared to a 96K insert.
3-it sounds(from your description) that you don't have everything routed thru the hardware, which is very likely to cause issues. Anything routed thru the hardware and mixed with anything NOT routed that way stands a very good chance of causing comb filtering at best(and out-right flamming at worst). A possible solution to this would be to route anything NOT going thru the SSL comp thru another hardware insert, even if its nothing but patched right back in(its likely the SSL hardware, being analog, has no inherent latency, so a straight wire would theoretically match the latency of the round trip; a combination of converter latency and buffer setting.
4-inserts on a MASTER track do not get ADC(which probably means my #1 above is moot).

Hope something here helps

[brunogermain]

Answering your questions:


1. Yes. My HW Inserts have 105 samples of delay, which translates to 1.09 ms at 96kHz and 2.18ms at 48kHz. I always make sure it's set according to the session's sample rate

2. When I open the session there is no latency. It happens after a while

3. The SSL is not routed through a HW Insert. The chain is: Master->SSL->Audio track IN.



The problem here is that the comb filtering is not constant. It happens randomly while playing back the audio. This happens ONLY working at 48kHz. The same mix converted to 96k sounds perfectly in phase