cliffr
Na, Cubase has been 64 bit for a while now.
I only know beacuse I was thinking of adding it to my tool kit a while back so checked that first.

You misunderstood what I was saying.  Certainly Cubase runs as a 64-bit application.  But the internal data stream for sound is, I believe, still 32-bit only.  You do not get a choice between 32 and 64 as you get in SONAR.  Now, I could be wrong about that because I've just recently acquired Cubase and am still reading a massive amount of documentation, but that's my understanding so far.

mike_mccue
Here is an interesting quote from the link that Super G posted:
 

... firstly, it means increased dynamic range for the whole system, making it pretty hard to overload the mixer even when working with a large number of tracks ...

Technically the first part of the statement is correct.  Double precision gives you increased dynamic range.  But the second part is bull, IMO.  The dynamic range of single precision floating point is already so much in excess of what would be needed in any real-life situation that there's no reason to make that the rationale for going to double-precision.  Less rounding error and more precise DSP calculation, yes that's a definite gain with 64-bit.  But offering more dynamic range than you'd need when single-precision wasn't up to the task, not really.

My OP question was a reaction to the fact that you don't have meters in the signal flow at the input of the summing busses and so you don't know what your head room is.

I think we've established that headroom at the summing busses is mostly irrelevant, at least on a purely technical level, because we're still in floating-point land there. 
 
What's more significant is that most of us don't have meters at the hardware output. Instead we rely on software that can report overs and interpolate where the signal will actually peak when reconstructed.
 
Even when provided, such information may not be accurate. Software cannot differentiate between an over and a legitimate sample value of 0dB, so it assumes that N consecutive samples at 0dB constitutes an over, with the value of N being up to the developer's/manufacturer's discretion. Consequently, hundreds of overs may not be reported. 
 
Whether you're working in integer or float, 16, 32, 48 or 64 bits, the best strategy is and always has been to just leave some frickin' headroom. The common practice of pushing peaks right up to -0.3dB or higher is totally unnecessary. 

bitflipper

My OP question was a reaction to the fact that you don't have meters in the signal flow at the input of the summing busses and so you don't know what your head room is.

 
 
Whether you're working in integer or float, 16, 32, 48 or 64 bits, the best strategy is and always has been to just leave some frickin' headroom. The common practice of pushing peaks right up to -0.3dB or higher is totally unnecessary. 

Thank you, oh wise one.  This is indeed the way of the tao.  Leave some frickin' headroom, will ya??  Your mastering engineer will thank you.  Many times over.

As I have already said one can very easily keep well clear of any limits of the digital system. What is even better about the K System approach is that for your final mix to finally end up at the reference level eg K-14 the buses are actually below that. The buses might be sitting down at K-17 for example because it is when they finally add up on the masterbuss the final mix comes back up to K-14. So our buses are sitting at an rms value of say -17 which means 17 dB of headroom above that for peaks. Pretty decent amount of headroom. Even a powerful transient might reach 12 dB above the rms value meaning the loudest peaks on my buses are only making up to -5dB at the most.
 
You don't actually have to monitor buss levels at all in fact. You only really need to monitor the final stereo buss level. Because you will never really go over your chosen ref level (on your masterbus) which is keeping the buses slightly below that.
 
You just turn up the monitor level now in your room to either 85 dB SPL or higher. The reason why people overload their tracks and buses is that they are keeping their monitor volume down too far and hence they are pushing everything so hard in order to hear the music. K System is about ref levels within your DAW AND monitor volume settings. I have a SPL meter permanently setup in my control room and it is very good to have it there. It tells you a lot about what is going on.
 
Even in 16 bit system if you choose a K-20 ref level and that is even lower, you still have 70 dB of dynamic range underneath at your disposal. At 24 bit this goes up to 120 dB below -20dB! No reason to be peaking anywhere up near 0dB FS is there!

 
Sometimes I wonder...
 
How does a simple technical question turn into a series of pedantic and remedial lectures about gain staging?
 
 
Goofy.
Slightly Frustrating as it seems condescending.
Goofy.
 
 
all the best,
mike

It's hardly pedantic if a point is made that renders the original question moot.

 
In my opinion, responding to the original question:
 
"Is Pro Tools native up to par now or does 64bit floating point mixing require the more expensive HD option?"
 
With a lecture about gain staging is simply stubborn. Describing the question as "moot" finalizes the disregard.
 
In all fairness you came very close to answering the question in post #2, but I had recognized some factual errors (namely that the mix engine is apparently 64point float and the efx routing is 32bit float.) and I still needed further info and or clarification to get those facts straight.
 
It was a simple question.
 
Taking that question, hypothesizing circumstances, and espousing a disregard for the actual and specific question is what it is.
 
I'm a big boy... I can sift through the chaff... but I wonder how helpful these replies would be to someone with less experience to guide them.
 
In my opinion the lecturing seems:
 
Goofy.
Slightly Frustrating as it seems condescending.
Goofy.
 
But, I like all of you guys a whole bunch... so please consider that I mean this in a good way.
 
 
all the best,
mike
 

Careful gain staging and metering = no internal clipping. Simple as that really. On any DAW. I have never clipped the AUX inputs in PT in my life. Why is that I wonder. And yet the tutorial the OP refers to internal clipping on the AUX buses. Ever wonder why it happened in the first place? Metering is not a workaround, it is a necessity.
 
You can walk along the road and fall into a hole and then you will need a lot of enery and effort to get out of it, or you can look ahead and just walk around it.
 
Mike I do respect your deep interest in these things and it often uncovers things we did not know and that is always a good thing. But for those of us who are being paid to pump out mixes every day that sort of stuff is the very last thing on our minds. There is just no time to think about it.
 
 

Yea...it's kinda like wanting to have a detailed understanding of quantum "membrane" theory before you go to Starbucks for fear you'll drive into another universe on the way there. Or is it?? :-)