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that wouldn't be a pitch "correction" plugin then that would be a pitch shifting plugin ;) Yes, PT has pitch shift capability. It's not the greatest (IMO)...but it works. with regards to where to go to learn how to use PT, this forum is great. Also, the manual has pretty much everything you'll...

Then you need this http://shop.avid.com/store/product.do?product=324175094474352

For the 001? http://shop.avid.com/store/product.do?product=324175094480128

nothing attached. When you listen back after you bounce to disk are you listening back on the same speakers and through the Mbox mini?

could be a hardware buffer issue. What's your buffer set at?

I would try updating any computer drivers (graphics driver particularly), hardware firmware, PT CS updates, etc. Even though it works fine in Cubase, it can still be a problem since PT uses different resources and the hardware differently. Also, if you have any antivirus (or other programs)...

it can be a multitude of things. your hard drives, your memory, firmware needing updated, prefs need to be trashed, PACE drivers needs updating, CS fix needs to be installed.... check 'em all.

what is your computer specs? Are you using a RAID drive?

oh I adjust the waves to time align all the time. And I don't think NYMorningstar was saying it's physically impossible to move them in the time line and get it to sound good. I think he also was saying you won't get a perfectly, 100%, identical, in phase signal. As you can see from your own...

not necessarily. Because of different room reflections/modes (and microphone frequency responses) you will see different representations of wave lengths in your DAW. They will look similar, but will not be 100% identical. the funny thing is, I think we're over complicating this. If the OP's...

that's just a myth...or was true for very few devices/programs back in the day. The math is much more complex whenever down sampling from any number. If you just threw away every other sample you'd have things to deal with like twice the extra upper-harmonic content captured in the 88200 sampled...

create a new track...add the room tone or VO recording.... :confused:

The extra hard drive space and system resources required for this, outweigh any benefits of a higher sample rate...IMO. It's just not worth it. You can make just a great of mix at 48kHz as you can at 192kHz.

mix down to whatever source you're going to. If you're going to CD, you'll have to use 44.1kHz. Most other things should work with 48kHz just fine. I like to play it safe and keep it at 44.1kHz...unless I'm doing video work (or someone requests 48kHz). My sessions all start at 48kHz, though.

are you workin' with a stereo track? try that Do you see Rewire loading up Acid when PT starts up?

page 247 of the 8.0 Reference Guide:

It's not a bug....it's because you can't use Elastic Audio with SDII files. You should be using WAV or AIFF files anyway since they're pretty much the standard and cross compatible between platforms.

And you're sure you don't have any tempo or meter changes sitting out there? from the manual:

should be able to. However, you'll be limited to only the first 8 channels of ADAT. Word clock should be fine too.

like moresound said....almost ;) I'm being picky, but there is a difference. What we have is three different things. A mono/stereo signal...a balanced/unbalanced signal...and a type of connector. All of these are matched up in different ways. You can have a mono signal that is balanced and...