Homemade hardware i/o DAW delay compensation...

[PapillonIrl]

What are some of the workarounds people are using for this in DAW's without HW delay compensation (most of them at the moment it seems) ?

For example when I want to parallel compress a drum mix (post analog console eq) and have no HW comps left, I'll often print a stereo drum mix and realign it in time to the original tracks. Then mash it with an all-buttons in UAD1176 or Fairchild. I think the extra A/D/A doesn't really matter in these kind of cases/


Any software out there that tests the delay time of your HW loopback ? Surely there is ?

I'm sure this time next year most DAW's will have it. It would be so handy for me to audition post-eq stuff and mults through software compressors in realtime.

Nathan

[Kiwiburger]

IMO, don't trust specs or software. Take a reference waveform - could be anything. Send it out of your DAW and back in again using the external device in Bypass mode. Record the delayed signal to a new track. Normalise both tracks so they have the exact same level, then flip the phase on the delayed track. Now, shift the delayed track backwards until you hear the two signals cancel out perfectly (or as good as they possibly can). You've now found the perfect delay in samples to compensate for the total combination of all latencies.

[thethrillfactor]

Originally Posted by Kiwiburger

IMO, don't trust specs or software. Take a reference waveform - could be anything. Send it out of your DAW and back in again using the external device in Bypass mode. Record the delayed signal to a new track. Normalise both tracks so they have the exact same level, then flip the phase on the delayed track. Now, shift the delayed track backwards until you hear the two signals cancel out perfectly (or as good as they possibly can). You've now found the perfect delay in samples to compensate for the total combination of all latencies.

It will never cancel out perfectly if that's what you are hoping for.

Just running it through the processor(comp and EQ) in bypass will introduce artifacts.

By the way PTHD does compensate for the HW delays.

For a multed drumbuss you do still have to shift the compensation by one sample forward.

[Kiwiburger]

It will never cancel out perfectly if that's what you are hoping for.

Which is why I wrote "(or as good as they possibly can)" .

The point is - I like to use my ears to find the best delay time, rather than use a software calculation or advice from an equipment manual.

I use external A/D and D/A (Lucid & Benchmark). I could look up the latency for each converter, and read off the latency of my soundcard, and the latency from the plugin, and that would only be the theoretical delay compensation to apply.

The original question was about software for calculating the delay. IMO - unless ths software sent a test tone through the entire chain, it wouldn't be accurate either. Because items like external A/D and D/A converters don't report back the latency to the host software.

Although if you record a cloned track, you could make some software that could identify the time difference between them by looking at the peaks and counting the number of samples between the same peaks.

I just think that if you can find the latency by using your ears (looking for the best nulling), you can be confident you've got it right.

[thethrillfactor]

Originally Posted by Kiwiburger

I just think that if you can find the latency by using your ears (looking for the best nulling), you can be confident you've got it right.

First of all when you are trying to mix fast and without thinking trying to figure out latency times is a drag.

Its part of the reason i got out of mixing in PTools and back on an analog console around 2000.

Since then Digi has implemented their HW latency correction on PTHD and it works really well with their own converters.

[Kiwiburger]

I'm lucky to use Cubase SX, which has full plugin delay compensation. I see the lack of this in Protools LE as the major reason to avoid Protools LE - it isn't "Pro". Even the freeware DAWs have full plugin delay compensation.

If you have a DAW with full PDC, then you only have to work out your actual system latency once and remember how many samples to offset when required.

[Fredrik]

What you can do is to follow the above suggestion, but instread of flipping the phase just use the scale (ruler) in your daw, then you can see exactly how many samples late the new track is (or ms if that what you have chosen in PT).

[stagefright13]

For my 002r I use the chart on this site and it is right on the money...

http://www.elvenminstrel.com/recordi...auxlatency.htm

[synthoid]

In addition to setting the latency compensation as described above, you need to figure out when the latency will change, i.e., what kind of change to software / hardware / path through the interfaces / etc will result in a different latency. I have a relatively simple computer set-up, but even through my RME fireface and into my apple I measure a couple different latencies based on settings of the software and interface and paths of the audio into and out of it. This stuff is such a headache - I absolutely detest this (latency and especially variable latency) about recording into computers. Whenever I track into my HD24XR I'm so much more confident about the timing. If you ask me, computers will be ready for prime time recording when there is ZERO misalignment of audio always. They shoulda started with that as the first requirement and built the silly DAWs up from there.

-synthoid

[Benmrx]

Originally Posted by stagefright13

For my 002r I use the chart on this site and it is right on the money...

http://www.elvenminstrel.com/recordi...auxlatency.htm

Thanks for the link. I'm gonna tape this up in my control room tomorrow.