Delay compensator ??

musurgio · Post by **musurgio** » Mon Apr 23, 2007 11:47 pm

I really would pay for that !!
Can this be done in SFP ?
We know that some modules have more latency than others.
For instance SPL is around 39 samples.
Most plugins are 4 samples some are more.
Can there be a mixer with latency compensation ?
Or maybe at least a Latency IDENTIFIER ?
Regards,
Dimitrios

tgstgs · Post by **tgstgs** » Tue Apr 24, 2007 12:29 am

take a sine tone and feed to the circuits you want to compare;
record these two or more waves and whatch with a editor;
then plugin a delay to the faster ones;
you only can adjust to the highest delay;
-------------

EDIT other translator makes other sense sorry

should be possible to automize between to defined points but calculations had to be redone after each changing of the circuit!

good vibes from vienna

musurgio · Post by **musurgio** » Sat Apr 28, 2007 12:58 am

Thanks for suggestion.
I have some methods calculating the latencies I just wondered if there can be made a plug that will identify any delay when inserted after and before a certain device.
Thanks
Regards,
Dimitrios

at0m · Post by **at0m** » Mon Apr 30, 2007 9:58 pm

tgstgs,
a saw wave has much easier to recognise x-crossings than a sine - one side is vertical :)

musurgio,
any delay compensation would just introduce extra latency in other channels, as you can do with some mixer's built-in sample delays, or with redmuze's phasefix which also allows you to accurately detect any delay.

musurgio · Post by **musurgio** » Tue May 01, 2007 12:07 am

Dear Atom,
Yes thats true but a delay compensator should automatically DETECT the latency that each plugin introduces and latent the other channels accordingly.This is how Cubase and other daws work.
I don't care if al channels will get a delay of some hundred samples in a mix scenario I only care about phase relationships !!
Rgards,
Dimitrios

Post by **garyb** » Tue May 01, 2007 12:48 am

don't use a neve....

hifiboom · Post by **hifiboom** » Fri May 04, 2007 10:30 pm

musurgio, why do you need sample accurate audio ?

I mean,
are you annoyed of hearing 4 samples delays on your mixes...?

musurgio · Post by **musurgio** » Sat May 05, 2007 12:29 am

Dear friend,
YES phase accuracy is a must when you do multi micing recording like drum recording !
When recording drums you could use almost 12-14 microphones !
ALL need to be phase accurate because then you will get flanging effects !
ALSO please note that one of the differences analog sounds better than digital is the phase relationship !!
Regards,
Dimitrios

hifiboom · Post by **hifiboom** » Sat May 05, 2007 9:58 am

ah okay I understand,

you place a mic for every single drum sound onto an extra channel...

So your problem is, if every mic on a solo drum sound is recording a bit of low level signal of the other solo drums?
So there will be some low level flanging/phasing effects..?

finally you even have to use the exact same cable length on each mic connection, else the cables will introduce latency, too.

astroman · Post by **astroman** » Sat May 05, 2007 12:18 pm

cable capacitance, inductivity, variance in the analog parts of each channel, and even the drummer himself has a deviation of more than 1/10.000 sec (4 samples at 44k) of 'latency' with every breath he takes...

cheers, Tom

Post by **hubird** » Sat May 05, 2007 1:25 pm

astroman wrote:, and even the drummer himself has a deviation of more than 1/10.000 sec (4 samples at 44k) of 'latency' with every breath he takes...

...not causing phase problems tho

you know those typical television talk shows, you can often hear delay related effects caused by the use of several mics.
I find myself often asking if it's that difficult to avoid.
There are gates, de-compressors, and -indeed- delay compensating hard- or software for when signals occasionally pass the gates.
Even the timing of the actual broadcasting time could be defined by that delay correction, to avoid a-sinc imaging).

Shroomz~> · Post by **Shroomz~>** » Wed May 09, 2007 10:32 am

Hi musurgio, there's a range of modules in the dsp module list within sdk which are called 'compensate delay X' etc. There's a few different ones & a linker module, but I've not had any luck getting them to work as they're not documented & don't have any obvious or apperent paramters for setting up compensation. When I checked them out (gave them a compensation job), they basically didn't seem to do anything. Must have to dig deeper.

musurgio · Post by **musurgio** » Wed May 09, 2007 11:15 pm

Thanks Shroomz for looking !!
I saw them I suspect that in the liinker ther is a MaxxD out.
Maybe there you have to define the maximum delay with some sort of delay module and maybe this will be taken into account when compensating.
I assume that you have to connect the "effect" out to the linker and the "no effect" to the compensate delay module.
Then maybe the delayed signal present at the linker will give the to the compensate module the information as how much to delay it.
So there can be probably an insert effect module where each time an effect is inserted there "must" be an output going pre-effect to the delay compensation module and post-effect to the lnker.
If that will not work then maybe this is to "manually" compensate ...
Hope these humble thoughts might help in some way !
The autonmatic delay compensation is a VERY VERY important aspect thats why it is implemented on big named daws like Cubase Nuendo,etc...
REgards and thank you for considering !!!
Regards,
Dimitrios

Shroomz~> · Post by **Shroomz~>** » Thu May 10, 2007 1:06 pm

Hi musurgio,

no problem, I actually looked into those modules a couple of months ago before getting sucked into the dsp management conundrum.

Let me know how you get on with those compensate modules!

Immanuel · Post by **Immanuel** » Thu May 10, 2007 1:52 pm

Here goes a DIY solution:

First: measure the delay of the devices you use. Write it down for later reference.

Second: Use Red Muze's 200 sample delay device - one on each channel. Set it to max. Each time you add a device to a chanel, you drop the delay of that channel by the number of samples, that the particular device delays your signal with.

This should be much easyer than adjusting everything else.

at0m · Post by **at0m** » Thu May 10, 2007 2:56 pm

Then you reload the project and the insert moves to another dsp, or worse, to another card, and your measuring goes all bananas. Good luck on this one :)

Shroomz~> · Post by **Shroomz~>** » Thu May 10, 2007 3:30 pm

That's the conundrum. I don't understand how we can guarantee sample accuracy without specifying specific dsp allocations for devices that span more than 1 dsp. I must be missing something.

spacef · Post by **spacef** » Thu May 10, 2007 3:35 pm

do a reclock of your board, normally, the devices (or parts) are reallocated on consecutive dsps and on board 0 by priority (ie, the master board). sorry, I use mainly spacef mixers and delays, but i have FP8, a p100 lite and an echo 35 in the aux insert slotsand project always reloads on same dsps. i tried with 2 boards. board gen I do not need reclocking

(press slave/master back and forth once in your samplerate setting). Too bad i didn't do the video for 2 boards (2* gen II boards) (sorry, talking by my own mixers here (FP8), i didn't try others).
nothing unusual or extraordinary as far as i'm concerned...

Shroomz~> · Post by **Shroomz~>** » Thu May 10, 2007 3:37 pm

Also, is it just me, or are inserts & insert devices no mans land in terms of sample accuracy?

edit... doh!! sorry, that's exactly what Atom refferred to !!

Shroomz~> · Post by **Shroomz~>** » Thu May 10, 2007 3:39 pm

Ah! Spacef, we were posting at the same time

Thanks for the info!