Is this normal 11R latency?

[miketeachesclass]

I'm testing latency, as I'm doing a session later.

I sent a pre-recorded click track out my 003 output 1 to a direct box, and plugged it into my 11R microphone input.

My 11R and 003 are connected in both directions via SPDIF. The 003 is the playback engine selected in PT9.

I record the pre-recorded click through the 11R mic input and come back with about 192 samples of latency in timeline placement. Is this normal?

[samplaire]

yes. Because it depends on many buffer settings that can give such result. Nothing special.

[miketeachesclass]

I thought this was one of the primary reasons for using avid hardware.

So even if my 003 buffer is at 64, I'll have latency recorded back?

I guess I must be confused at the best way to record with no latency on the 11r in the same way you can with the 003.

Can anyone shed some light on this?

[samplaire]

AFAIK the only 0 latency hardware is HD (TDM for older systems)

[miketeachesclass]

The LE hardware has been sample accurate (with the exception of the 8.0.4 release, and the driver with PT9)

I'm just trying to figure out the best way.

[miketeachesclass]

What this amounts to is that if you use the 11R, send the rig outputs via spdif, and the dry signal via analog, there is a different amount of latency for each.

This must be problematic for someone else, right?

[mykhal c]

Originally Posted by miketeachesclass

The LE hardware has been sample accurate (with the exception of the 8.0.4 release, and the driver with PT9)

I'm just trying to figure out the best way.

do you get the same latency when you just use the SPDIF input on the 11R into the 003R?? IOW take the roundtrip out of the config where it goes out of the 003 then into the 11R and back again ?

[miketeachesclass]

Originally Posted by mykhal c

do you get the same latency when you just use the SPDIF input on the 11R into the 003R?? IOW take the roundtrip out of the config where it goes out of the 003 then into the 11R and back again ?

I guess what I'm thinking is this should be the same basic test at the 003 latency test;

Taking the analog out of the 003, sending to the 11R analog input.

(if we took this analog output of the 003 and plugged it into an analog input of the 003, we should get effectively no latency because PT compensates for the converter delay)

It seems like if things are tracking properly, I should be able to achieve no latency from the 11r to the 003 via SPDIF.

In fact, I've gotten different numbers. In one session at 48k, I got 192 samples. I closed it and re-opened... same session... now I get 210 samples.

I'm supremely confused.

[mykhal c]

Originally Posted by miketeachesclass

What this amounts to is that if you use the 11R, send the rig outputs via spdif, and the dry signal via analog, there is a different amount of latency for each.

This must be problematic for someone else, right?

Key and i had a long conversation about this today. there is definitely some funky stuff happenin'. if you take an AUX track and change the output selection you'll notice delay compensation show up (if you have it turned on). i've done soooo much testin' lately that i'm kinda tested out but the current drivers for PT9 which are the same as 8.0.4cs2 still need a bit of tweakin', at least for my 002 console. for instance when i send out via ADAT to my Digimax and back in ADAT via the Digimax then my audio is early (forget how many samples right off). anyway, all that to say, do a test at your usual bit/sample rate and each buffer size to get a feel for what you're gettin'...

[miketeachesclass]

Originally Posted by mykhal c

Key and i had a long conversation about this today. there is definitely some funky stuff happenin'. if you take and AUX track and change the output selection you'll notice delay compensation show up (if you have it turned on). i've done soooo much testin' lately that i'm kinda tested out but the current drivers for PT9 which are the same as 8.0.4cs2 still need a bit of tweakin', at least for my 002 console. for instance when i send out via ADAT to my Digimax and back in ADAT via the Digimax then my audio is early (forget how many samples right off). anyway, all that to say, do a test at your usual bit/sample rate and each buffer size to get a feel for what you're gettin'...

Glad I'm not crazy.

We should be expecting this to show up properly, right? I mean, the fact that it's either early or late is a driver problem, not just "the way it is"?