Topic: New Sound Devices MixPre II-series coming?

[jerryfreak]

i use soundforge. im sure audacity can do it. I just select the two tracks and copy them to a new project, then save

[voltronic]

I believe Reaper handles poly wav files also, thought I haven't tested it myself as I don't own one of these recorders, and I don't enable the poly file option on my Zoom F6.  I can say for sure that stereo 32-bit float files work fine.

[morst]

Geez. So, I’m that case, can audacity handle the poly wave files?  That might be easier.
Import poly > highlight left and right ISO > stereo pair. ?

Once you drop them into Audacity, they arrive as mono tracks.
Then you can click the Little Tiny But Incredibly Important Triangle on each left channel, and select Make Stereo Track for each pair.

[justink]

Geez. So, I’m that case, can audacity handle the poly wave files?  That might be easier.
Import poly > highlight left and right ISO > stereo pair. ?

Once you drop them into Audacity, they arrive as mono tracks.
Then you can click the Little Tiny But Incredibly Important Triangle on each left channel, and select Make Stereo Track for each pair.

Boom. This is what I was thinking. So it is pretty easy?  Thanks.

[morst]

Boom. This is what I was thinking. So it is pretty easy?  Thanks.

It's pretty simple.

The only way it could be easier would be if Audacity made it look more Mac-like on a Mac, or more PC-Like on a PC, but that doesn't seem to be the plan.
Audacity lets you assign keyboard equivalents, I have not looked to see if you could make one up, then select the left track and then type your special command?

[noahbickart]

I believe Reaper handles poly wav files also.

It does.

For my workflow, wave agent was really useful. Using my macbook pro, I would point wave agent to the master files through usb-c to the the mixpre6. Then I'd create untouched, full show, stereo files for each microphone pair. Then I'd import those files into reaper.

I liked having an easy way to create a master raw file which would eventually be archived. It's a little cumbersome to use reaper, in which you have to do multiple renders to create such files.

Here's a little MacOS app that gets me much of the way there (only problem is that this can only split polywave files to mono):
https://public.msli.com/lcs/audiomove/

[GLouie]

I use both Waveagent then Reaper, or Reaper alone for polywav files from the MixPre10T. It just kinda depends on how you like to work.

Reaper 5's obscure way to open the tracks of a polywav file is to:

-open the polywav, you see one "track" with your multichannel polywav.
-go to the "ITEM" menu
-select "Item Processing"
-select "Explode multichannel audio or MIDI items to new one-channel items"

The channels of the polywav will "explode" like a subset of individual channels with full controls. Note that is isn't actually separating out the tracks, but you can work and process like it did.

[justink]

I sent this question to Paul a day or so ago but in the event he doesn't check his messages, anyone know the answer to this?

Running 32bit, and assuming set up is like this:

In "System" set the mode to Custom.
In "Custom Setup" set "Gain" to "Basic" and "Channel" to "Advanced."

Link 1-2, 3-4, etc...

1.) Is 32bit mode just a fixed gain regardless of how you play with the knobs?

2.) Is it possible to adjust the gain manually on ISO tracks?

My question is because usually when running 24bit, I still like to get as close to +0db because I felt like my recordings always sounded more "full" than peaking around -12db and then raising in DAW.  So, I guess my question is, does it even matter where I set the levels in 32bit?  Or would I still want to run them hot?

[Paul Isaacs]

1) In 32-bit float mode, you still have adjustable gain if you want to use it.
2) Yes you can still adjust ISO gain in the mode you describe. You have up to 96dB of gain for each ISO using the knobs.
It doesn't really matter where you set the levels other than setting them so you can hear them when monitoring in your HPs.

[voltronic]

1) In 32-bit float mode, you still have adjustable gain if you want to use it.
2) Yes you can still adjust ISO gain in the mode you describe. You have up to 96dB of gain for each ISO using the knobs.
It doesn't really matter where you set the levels other than setting them so you can hear them when monitoring in your HPs.

Hi Paul,

A quick question related to the discussion on the F6 thread:

Is the gain in the MixPre-II pre or post-ADC?  I looked for a block diagram and could not locate one.

[Paul Isaacs]

Post-ADC.

[aaronji]

^ I assume that is for 32-bit mode and not 24-bit mode (why would you need analog limiters otherwise)?

[voltronic]

Post-ADC.

Thanks.  So is there a nominal fixed amount of gain the signal goes through in the analog domain before it hits the ADC?

^ I assume that is for 32-bit mode and not 24-bit mode (why would you need analog limiters otherwise)?

Related question - is the gain stage applied to slightly different places in the chain depending on if it is in 24-bit fixed or 32-bit float mode?

[aaronji]

Post-ADC.

Thanks.  So is there a nominal fixed amount of gain the signal goes through in the analog domain before it hits the ADC?

If you look at the diagram in the patent application, it shows separate analog gain/filter stages prior to each ADC ("lowest gain stage", "mid-gain stage" and "highest gain stage"). Maybe the actual implementation differs in some way, but I think it is not unreasonable to assume that is how it works in the MixPre IIs.

^ I assume that is for 32-bit mode and not 24-bit mode (why would you need analog limiters otherwise)?

Related question - is the gain stage applied to slightly different places in the chain depending on if it is in 24-bit fixed or 32-bit float mode?

I was explicitly told by a Sound Devices tech that, in the first generation MixPres, there was an analog gain stage at the pre-amp and a digital gain stage after conversion (i.e. analog gain, or trim, and digital fader). Perhaps that is different in the second generation, but I doubt it. The specs list the mic input overload as "+14dBu (limiters on or off)", which is unchanged from the first iteration. Analog limiters don't really make sense, at least to me in my non-audio engineer brain, if they have no effect at the input. Maybe I am missing something, though...

[Gutbucket]

If you look at the diagram in the patent application, it shows separate analog gain/filter stages prior to each ADC ("lowest gain stage", "mid-gain stage" and "highest gain stage"). Maybe the actual implementation differs in some way, but I think it is not unreasonable to assume that is how it works in the MixPre IIs.

That's how a dynamic range greater than that of any one ADC alone is accommodated.  The limited input range of each ADC is identical, so the signal is split and routed to each with different amounts of pre-amplification, then analyzed and recombined which is the special sauce.

I was explicitly told by a Sound Devices tech that, in the first generation MixPres, there was an analog gain stage at the pre-amp and a digital gain stage after conversion (i.e. analog gain, or trim, and digital fader). Perhaps that is different in the second generation, but I doubt it. The specs list the mic input overload as "+14dBu (limiters on or off)", which is unchanged from the first iteration.

 
Analog gain stages for signal matching to the requirements of the ADCs, yet not necessarily featuring adjustable trim (by the user or even by the system).  That detail might be a source of confusion here.

Analog limiters don't really make sense, at least to me in my non-audio engineer brain, if they have no effect at the input. Maybe I am missing something, though...

I agree, as long as the 3 ADC scheme is capable of sufficient range on its own, which seems the entire point of the doing all this.