Is there any sound degradation when turning down a master fader?

[Bob Olhsson]

There was a problem with this that got fixed something like twenty years ago. I remember automating the faders on waves ren Eqs because it was by far the best sounding fader on any DAW.

Those were the daze...

[Craig F]

Coming at it from the Post perspective riding the Master usually messes up your room calibrations & hardware meters

[bad jitter]

Originally Posted by JFreak

There is a white paper available on 48bit mixer (TDM) but the principle should be the same on native.

Not really: http://www.soundonsound.com/sos/jun1...es/pt_0610.htm

[dr_daw]

So, I'm revisiting this post this evening and I have a question for the OP; What are you doing to get

I have practiced this method and gotten quite good at it...

?

[scrnplyr]

Originally Posted by Carl Kolchak

You can test this by creating a 32bit floating point session, cranking up the master fader by 12dB (so that your audio distorts), and printing it to another audio track.

The new recording will be horribly distorted, but if you open the Audio Suite gain plug-in, reduce the gain by 12dB, and process the file, it will be restored to it's former glory.

You can test that by playing your original source audio file, and the newly processed file together, inverting the phase polarity of one against the other - they will cancel out.

I haven't had to do this in a session yet, but it blows my mind to realize its possible. Does this concept cross analog lines? In other words if I create a distorted analog mix +12db record it to tape and then import it back to digital could i fix the over modulation by reducing the gain 12db?

[Carl Kolchak]

Originally Posted by scrnplyr

I haven't had to do this in a session yet, but it blows my mind to realize its possible. Does this concept cross analog lines? In other words if I create a distorted analog mix +12db record it to tape and then import it back to digital could i fix the over modulation by reducing the gain 12db?

Unfortunately not - the only chance you would have of recovering an over-modulated analogue recording / severely clipped fixed point digital recording, would be to either redraw the waveform by hand (after using the Audio Suite gain plug-in to reduce it's overall level), or to get the best result you could with something like Cedar / RX.

The only reason it is possible to recover the clipped digital audio I mentioned in my previous post, is because I was specifically referring to a floating point system.

I suppose it could theoretically be possible to record to a 48bit fixed point file, whilst clipping a 32bit floating point mixer, and still recover the audio, provided you didn't exceed the dynamic range of the 48bit fixed point scale.

[tj_davies]

Originally Posted by scrnplyr

I haven't had to do this in a session yet, but it blows my mind to realize its possible. Does this concept cross analog lines? In other words if I create a distorted analog mix +12db record it to tape and then import it back to digital could i fix the over modulation by reducing the gain 12db?

Uhm, nope. No. Naah. Uh-uh. Niet. Nein. Non!!!!

[CME]

Originally Posted by Carl Kolchak

(snip)

I suppose it could theoretically be possible to record to a 48bit fixed point file, whilst clipping a 32bit floating point mixer, and still recover the audio, provided you didn't exceed the dynamic range of the 48bit fixed point scale.

I don't think that's possible. Floating point is the key here and what allows ridiculous theoretical headroom. I don't know the math, but I'm pretty sure 32-bit floating point has a lot more headroom. And that's what allows the previous situation of recovering a clipped 32-bit float file. In fixed point, it you clip you clip. Much like analog.

And fwiw PT has had a 64-bit float mixer since v9.

[tj_davies]

Originally Posted by Carl Kolchak

I suppose it could theoretically be possible to record to a 48bit fixed point file, whilst clipping a 32bit floating point mixer, and still recover the audio, provided you didn't exceed the dynamic range of the 48bit fixed point scale.

I think the question is, if it works in the analog world, so that is a definite no.

Timothy

[Carl Kolchak]

Originally Posted by CME

I don't think that's possible. Floating point is the key here and what allows ridiculous theoretical headroom. I don't know the math, but I'm pretty sure 32-bit floating point has a lot more headroom. And that's what allows the previous situation of recovering a clipped 32-bit float file. In fixed point, it you clip you clip. Much like analog.

I agree, logically, you'd expect that if the floating point mixer is clipped, then a fixed point audio file would just be a document of that clipping - but I'm pondering theoretically - if it were possible within an enclosed environment like Pro Tools, to clip a 32bit floating point mixer by 12dB, whilst generating a 48bit fixed point audio file (which is well within the dynamic range of a 48bit fixed point scale), it might seem reasonable to be able to attenuate that file by 12dB, and be back within the "clean" zone of the 32bit float mixer upon playback.

I don't think that would actually be the case, for the obvious reasons - but as it's something I can't actually test in the real world, it's just a thought experiment.