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Hello, I have a few pieces of outboard gear that I'd like to incorporate into my mixes. I know that working with plugs and outboard at the same time can be a headache, but i don't see any reason why it would be too hard, as long as a latency plug is inserted on all tracks going out of the box.

Since DP lacks a "ping" feature, how do all you that use DP mixing plugs and outboard gear calculate expected latency? I have found that I need to set "Fine tune audio I/O timing" to 57 samples in order for overdubs to line up in phase. Do I simply add these 57 samples to whatever buffer setting I am working at to calculate the amount of latency bussing a track out to an EQ and then back into DP would create? Are the two things completely unrelated?

I guess i could sit down with a pen and paper and draw out exactly what was going on when i send things out of DP and then back in, but I'm interested in hearing how some of the members here with experience in this go about with their workflows.

thanks!

Hello, I have a few pieces of outboard gear that I'd like to incorporate into my mixes. I know that working with plugs and outboard at the same time can be a headache, but i don't see any reason why it would be too hard, as long as a latency plug is inserted on all tracks going out of the box.


I am using several outboard processors with DP. Here is how I do it:
I have a 2408 (mkII) and a 1296 interface. The 1296 has 12 analog ins and outs, and several outputs are connected to the inputs of external processors. So, in DP I enable a send in the fader window by choosing, for example, 1296 output 11 which feeds the input of a Lexicon MPX 100. I can send the signal through the 1296 out to the Lex and return it to my mixer (in my case two Tascam DM24s). This way I can automate the send to the unit. Latency has never been a problem this way. I can also do MIDI dumps into DP to save setups from the external processors.

I set up mine with manual tests using a recorded click sample.

At a given sample rate and buffer it's always the same so you just need to do it once for each setting you plan on working at.

Take the recorded audio, bus it out on an aux to a D/A and wire directly back into an A/D convertor and record on a new track.

Zoom large, compare offsets with the counter set to samples and voila, your own personal magic formulae.

I should add that if your summing into an external device such as a digital mixer;

Run two paths off the original "click" track.
1- Straight to the board and out the mix outputs and back into DP hard panned left
2nd - routed as you will use the analog patching then to board and out mix output hard panned right.

compare those files for the offset. In this case it doesn't matter how you return to the DAW (via A/D, light pipe, AES/EBU or SPDIF) as long as both paths use the same method.

I am actually summing in DP. If i was using a console/mixer/sumbox then there wouldnt really be a problem. But I want audio to go out of dp, get processed by my eqs, compressors, and reverbs, and then go back into dp. I have figured out the offset for sending audio out of dp, and then printing it back to disk. This is dandy, but I really need auxes and returns set up in DP that correspond to the outputs on my converters. Latency is introduced during playback, however, when bussing a track out of DP, through a processor, and then back in, in realtime. Some sort of latency compensation plug has to be strapped across the sends.

I remember a post somewhere by Kubi that explained how he set up a V-Rack with auxes in/out and used some sort of latency plug.

I normally wouldn't bother, and stay in the box, but there are only so many plugs i can run on my UAD cards, and I have these compressors and eqs sitting here, might as well use them. besides, if i can figure this part out it might actually be a fun change of pace.

Well then here you go:


http://www.collective.co.uk/expertsleep ... fixer.html


Works fine for me. I'm PPC though (Dual G5 2.3). If you're intel I don't know if it'll work........

neat! no im still on ppc.


so i just set the latency on this plug to:

buffer size + Value in DP's "Fine tune audio I/O"?


well ill have to give it a try.

thanks!

Please post your results!!!

so the first go at it with the Latency plug wasn't all that amazing. I recorded a drum loop (with my trusty alesis sr-16, 17-year old samples and all) to an audio track. I then duplicated this track, and bussed it to an output on my 828mk2 (Analog 1), which is connected to a compressor. The compressor's outputs are connected to the (Analog 3) input on my 828. So I set up an aux track with Analog 3 as an input, and input monitoring enabled. So it looked like:

DrumLoop---------------->Main Outs
DrumLoop Copy--------->Analog 1 (compressor in)
Analog 3 (compressor out)----->Aux 1------->Main Outs

I muted the No-latency monitoring in CueMix. When i hit play, there was obvious flamming, to be expected. I then inserted the Latency Fixer plugin on the DrumLoop Copy, but could not for the life of me find a setting that sounded ok. I tried setting it to the buffer setting, 1024, half that 512...I set it to the value I have set in the "Fine tune audio i/o" setting, which is 58 samples, and still no dice. 512+58, no go.

I'll reassess the situation and try again tomorrow. I don't NEED to get this to work, but i sure would like for it to.

do the test I suggested in my first reply and use that number.
With what you're doing you have ;

1- latency at the D/A output stage

2- latency at the A/D re-entry stage

and

3- any system specific I/O fine tuning required.

The test I outlined accounts for all of this.
Do the test in a project with no other tracks or plugins, Just the recorded click, snare hit or whatever and write that number down. It will always work flawlessly for any routing through those convertors at the sample rate and buffer you test it at.

thanks THW, I will try taking that measurement today. So I just record a click to a track, bus that track to an aux that outputs to an output on my interface, patch it directly to an input on my interface, and hit record? Should I set "Fine tune audio I/O" to zero? Does enabling "Input Monitoring" change anything?

thanks again

OldTimey wrote:thanks THW, I will try taking that measurement today. So I just record a click to a track, bus that track to an aux that outputs to an output on my interface, patch it directly to an input on my interface, and hit record? Should I set "Fine tune audio I/O" to zero? Does enabling "Input Monitoring" change anything?

thanks again

That's all you do.
I'd leave fine tune audio wherever it was that you've had it in the past.

Don't know about the Input monitoring part.
When I use my external stuff I just bring it back on an Aux track routed to the mix bus.


There's also this VST two piece plugin set that calculates it automatically. All you do is put one on the send track and one on the aux return and it calculates the difference automatically.
It does require use of a VST to AU wrapper.

Thanks again THW,

so input monitoring does not need to be enabled to hear aux inputs live...


im going to get to work on this as soon as i can!

That's thing about UAD cards...once you try a plugin that emulates how real hardware operates and sounds (and looks) you start wondering how cool a real 1176, or la2a might be...(not that i have those pieces...yet)

Very cool.

I was told once that some DAW's ping the outboard device for latency and auto set it. Is that true? "Wouldn't it be nice" (Beach Boys)

Timeline wrote:Very cool.

I was told once that some DAW's ping the outboard device for latency and auto set it. Is that true? "Wouldn't it be nice" (Beach Boys)

My other DAW (the "evil" PT HD 7.x with 192 I/O) automatically adjusts the round trip latency and let's you assign pairs of D/A-A/D convertors into named devices that show up as a plugin in the mixer right alongside the standard plugins.

I have my neve EQ's, a Drawmer 1960 and a 4 channel Bystrom tube compressor all hard wired in and named as such in my plug in list.

I wish DP would embrace that concept at least when using their own hardware. It's not as if they don't know what latency a 2408 mkIII introduces and allowing output/input at the insert point couldn't be that hard.........