Skip to main content
Participant
Jul 30, 2020 9:20 PM UTC
Question

Adobe Audition - Issue with 32-bit float Recording

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!

This topic is closed to new replies. Start a new post to keep the conversation going.

14 replies

SteveG_AudioMasters_
Community Expert
Community Expert
Jul 31, 2020 9:57 AM UTC

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.

Translate
Chicago Slingshot
Participant
Jul 5, 2021 1:55 AM UTC

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

Translate
chrisw44157881
Inspiring
Jul 7, 2021 3:34 AM UTC

Like I said, they don't explain it very well, and the last five words are very significant, especially 'additional'. If you have an analogue signal fed into analog electronics (ie, a mic preamp) it will have a performance ceiling at which point it will distort - and if you digitise that, you are digitising distortion. The best input stages you can build are limited by the Laws of Physics to have a dynamic range of about 120dB, and that is easily digitised in its entirety by a 20-bit ADC, never mind a 24-bit or 32-bit Floating Point one. That's not the issue. The issue is that the input is still overloadable by a mic if the gain is set too high - because the mic output from a loud source will, unless the gain is adjusted accordingly, overload the input and cause distortion. There's no such thing as a 'native maximum spl' either - all mics are different and there are many professional ones around that will produce undistorted outputs of levels that your ears could never stand. It's not the limits of the mic you will hit, but the limits of the input stage it's connected to. And that is what Sound Devices don't describe adequately. Their mic preamps are good - I use them - but they aren't perfect and they very much are oveloadable.

 

In a previous existence I spent a lot of time designing mic preamps. I'm reasonably confident that I know a lot more about it than you do...


each mic does indeed have its own listed as rated max spl. you would be hard pressed to find a mic that can withstand >142db (mix-pre's specs)

 

you can theoretically cram more dynamic range into 32 bits,  per spec IEEE-754 supports up to +770 dBFS. Does that make the mix-pre II not a true 32 bit recorder then?

 

I think the idea here is file format. you can store a special kind of data file format("audio signals above 0 dBFS are preserved in the file"), that is compatible with your computer to recover. if you record at 24 bit, it simply clips...

 

If you want to say that 32 bit doesn't exist because the mic and recorder can't fulfill 770 db, then well, I can't convince you otherwise.

 

Translate