I am using Ozone3 with the winamp VST bridge.

With most 16 bit music there is little headroom left for EQ or DSP affect. When you add them you start clipping. I have a 24bit DAC. Is there any way to have ozone convert the 16bit to 24bit which would give more headroom to add the affects?

Converting to 24 bit (or floating point, which is better) will not in itself add any headroom; the differences are all at the very lowest signal levels. You can, however, first de-amplify the file to add headroom for your other processing. You can do this in 16 bit format, but the quantization errors will be much smaller if you first convert. If you do work in 16 bit, dithering the transforms, including the negative amplification, is recommended.

To my knowledge Winamp is limited to integer processing, which makes it prone to clipping. Using foobar2000 would solve the problem as it uses 32-bit floating point processing, which makes clipping impossible inside the player. Of course the signal still has to be attenuated if it contains samples above the 0 dB mark. But this can easily be done by adjusting the preamp setting or by using an additional DSP plugin.

actually nevermind, i'm a twit

glenda: what he ^ said is exactly it. when you convert to 32 bit float you can "recover" peaks that are over 0 dBFS in the mp3 stream. you can then throw a limiter in the chain to ensure that there will be no clipping. with foobar you can do this manually or on the fly whichever suits your needs but not with winamp.

Ozone 3 uses 64-bit processing. I'm not sure how the 64-bits are "calibrated", but I would expect this to give you essentially unlimited headroom. (But, I don't know about the Winamp VST bridge.) As long as you normalize before saving to 16-bit, you should be OK.

I use GoldWave. It uses 32-bit floating-point for processing and temporary storage. As long as I normalize (GoldWave calls it "Maximize") before saving, the file is never clipped by the processing.

Unless you are trying to repair clipping by rounding-over the peaks, there isn't much point in restoring the peaks and then re-squishing them with a limiter...

Peaks in mp3 can be intact even when 100dB over 0dBFS. mp3 is "floating point" too.

Of course, you'd have to use some specific tools to get an mp3 that wrong, but common compression pushes peaks about 0dBFS without introducing added clipping - that only happens when you play such peaks back without doing something about it in a float point / extended headroom domain.

Cheers,
David.

I only use flac/shn, no MP3's. My concern with using the preamp of Ozone is that effectively reduces resolution by 1 bit per 6 db. Unless the "32 bit" processing somehow gets around this.

I am think along the lines of those digital volume knobs CD players used to have, lower volume=lower resolution. Isn't a VST preamp the same?

Yes, lowering the volume will reduce the resolution if working with integer values. But using 32-bit floating point you will always have the same resolution (24-bit). That's one of the two benefits (the other being clipping-proofness).