Who else doesn't get how different input and output buffers shall work?

[nst7]

As mentioned, it should be the same as tracking thru Native plugins before. But this feature will just make it easier to do so, and be able to do it with more tracks and plugins going on before you finally hit that point where you need to deactivate some plugs, or bounce down to stems, before continuing to do so.

[nigelpry]

My impression that the concept is to allow low latency on tracks you are recording on, so the musos don't get a noticeable delay in their cans. This is NOT about conquering latency when trying to use plugins on the track you are recording on, that would require warping the space time continuum ;-)

When doing overdubs, you still want low latency for a noticeable delay free headphone mix on the tracks you are now going to record on.

But by then, the tracks you previously recorded may have lots of plugins instantiated, you might have effects busses, and you might have lots of virtual instruments playing back too.

It's about pro tools dynamically creating a bigger playback output buffer for that previously recorded data, so your computer can cope better. So it will be manipulating the playback timing of the previously recorded material, while at the same time maintaining very low (unnoticeable) latency on the tracks you want to record on now.

The only logical way to do that is to playback the previously recorded material early, by precisely the correct amount of samples so that it appears at the output buss at exactly the same time as the recording track spews its audio back out having passed through the track.

If you try to instantiate plugins, or convoluted routings on the track being recorded on, then there will be extra latency building up due to the delay caused by those plugins/routing. There's no way of avoiding that. Some simple plugins can be used, just like they can now, but anything that generates noticeable delay won't be of any benefit ... it will just put the musician off.

[DonaldM]

I frankly don't care HOW it works, only that it WILL work. They could be using a mini Harry Potter waving a magic wand inside my PC for all I care...as long as I get low latency on the input without sacrificing the output.

[Raoul23]

If this new buffer works as good when recording with plugs as Avid say (or not say as Avid didnt mention tracking through plugs and the low latency) people wouldn't pay the big cash for HDX because apart from more I/O what other feature do you really get with HDX compared to Native. Id really like to see this working and what the limitations are before I upgraded my TDM rig.

[nigelpry]

Avid is not talking about miraculously achieving low latency when tracking through plugins that themselves cause extra latency ...

When you overdub it is always a compromise ... how low can you set the buffer to give a delay free headphone mix to the musician while not having pro tools fall over with errors because of the cpu load of all those previously recorded tracks loaded up with plugins, and virtual instruments etc?

The heavier the load of previously recorded material and the pulgins you've put on those tracks, the higher you have to set the buffer, at the moment (in PT10 and earlier), and that then starts to affect the musician because he can hear a delay in his headphones for the stuff now being recorded. ...

Well this is Avid's way of trying to resolve that problem. I can't wait to try it out, it'll be one of the first new features that I take for a spin!

[Raoul23]

Wouldn't this feature be stepping on the toes of what HDX has to offer

[mykhal c]

Originally Posted by nigelpry

Avid is not talking about miraculously achieving low latency when tracking through plugins that themselves cause extra latency ...

When you overdub it is always a compromise ... how low can you set the buffer to give a delay free headphone mix to the musician while not having pro tools fall over with errors because of the cpu load of all those previously recorded tracks loaded up with plugins, and virtual instruments etc?

The heavier the load of previously recorded material and the pulgins you've put on those tracks, the higher you have to set the buffer, at the moment (in PT10 and earlier), and that then starts to affect the musician because he can hear a delay in his headphones for the stuff now being recorded. ...

Well this is Avid's way of trying to resolve that problem. I can't wait to try it out, it'll be one of the first new features that I take for a spin!

which i actually did explain in my 1st post from a general programming approach that is...not necessarily Avid's way of attackin' the problem.

[Sardi]

Exactly.

We're all just repeating ourselves now.

[mykhal c]

Originally Posted by Sardi

Exactly.

We're all just repeating ourselves now.

yep...exactly what threads are for!!

[Shan]

Originally Posted by nst7

As mentioned, it should be the same as tracking thru Native plugins before. But this feature will just make it easier to do so, and be able to do it with more tracks and plugins going on before you finally hit that point where you need to deactivate some plugs, or bounce down to stems, before continuing to do so.

Exactly! It's no different than how it works currently. What's new is a dual buffer engine instead of our current single buffer. Function wise it remains the same as it always has.

We've had this problem along with it's solution in analog studios before computer recording and the DAW even existed. The concept and it's solution is nothing new since the days of tape.

Delay(latency) between the record head and repro head on tape = High buffer
Musicians going into the console = Low buffer
Monitor routing in the console = Solution

As far as "how" it works, in laymen's terms it's just routing under the hood(technically more to it than that when it comes to programming of course). Certainly not the oldest request for a multibuffer engine, but a more basic description.

A high H/W buffer gives us tons of CPU overhead, plug-in instances, VI's, large sessions etc, but impossible to recod without noticeable latency(It's great for mixing). A very low H/W buffer gives us extreme low latency(great for recording) but with a big hit in CPU, hence fewer plug-in instances, VI's, tracks, smaller sessions etc. The ideal solution is a multibuffer engine in which playback tracks stay at 1024, while only record tracks and live MIDI tracks are at 32(or the user defined settings). The result is a massive increase in performance in the many areas mentioned above in both native and HDX. As to "how" it works and programmed in the code, see analogy above.

Many of the DAW problems can model and echo the same solutions found in the analog studios of yesteryear. History likes to repeat itself, along with our posts.

Shane