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fred_joseph
New Participant
July 30, 2020
Question

Adobe Audition - Issue with 32-bit float Recording

  • July 30, 2020
  • 1 reply
  • 6459 views

I'm having this issue with recording audio in 32-bit where the peaks of the signal that exceeded 0dB are still being cut off when I reduce the gain after recording. I'm not being able to get the headroom that is expected when recording in 32-bit.

 

The reason why I'm having high +0dB peaks is because I'm transferring audio from a Tascam 388 and some of that audio on the reels was recorded a bit in the red. I'm running from the Tape Outs on the 388 and have the input gain at minimum level on the interface that I'm using for transfer (Tascam 16x08) so I can't reduce the signal input volume any further.

 

I'm recording the digital audio into Audition in 32-bit at 48kHz and again, when I try to recover the peaked audio I just get flat, cut off waveforms.

 

Thanks in advance for any replies!

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1 reply

SteveG_AudioMasters_
SteveG_AudioMasters_
Adobe Expert
July 31, 2020

You need to attenuate the input signal more. 32-bit FP is very essentially a software fix, and doesn't have any effect on the digitising part of an interface at all; if you overload that, it distorts at this point and there's nothing you can do subsequently to fix it. Your digitiser works as a 24-bit integer device - they all do. So if your input level goes above +8dBv, it's going to distort. What you need is in-line attenuators for the feed into the Tascam. I have to say that most devices with a line input will handle rather more signal than the Tascam does - I would have said that to deal with line level signals from any sort of pro gear, it should really be able to handle +24dBv, not just 8! If you can find attenuators that reduce the signal level going in by 10dB, you should be fine.

Chicago Slingshot
Chicago Slingshot
New Participant
July 5, 2021

I'm having the same problem.  I recorded some test footage on an iphone 12 that was intentionally clipped.  Then I imported the footage to adobe premiere pro and used the properties feature to confirm that it IS a 32-bit file and so is the sequence and project.  I then right clicked the audio to open it in adobe audition.  I lowered the volume of the clip and it remained clipped and distorted with a flat wabeform, just lower in volume.  Am I doing something wrong or is there a reason I cannot recover the peaks?  Thanks

SteveG_AudioMasters_
SteveG_AudioMasters_
Adobe Expert
July 5, 2021

Thanks for the reply.  I thought 32-bit float can record up to 700 db over 0.  And you can just bring it back down in a daw.  I've seen videos of people doing it.  I have concert footage that I would really like to save but having same problem with that.  


You need to understand that recording bit depth has nothing whatsoever to do with distortion. You can't fix distortion. Nothing can fix distortion. And Floating Point processing is only a software trick - real hardware produces integer-based outputs and they have an absolute ceiling of 0dB before they overload. Converting them to Floating point after this (which is the only way you can do it) makes not a shred of difference to that. Floating Point is not a recording format.

 

Yes you can process material, once it's in Floating Point, so that it massively overloads and can subsequently be bought back down without loss. But if that signal is distorted in the first place, then it will be just as distorted as it was when you return it to a more sensible level. So if you overload a mic stage and it distorts, you're stuck with it.

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