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converting 32 bit file to 24 using sampler?


GuyBorlander

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hi! 

there is this great bass sound from a sample pack I have found bit it is "32 bit" when I drag it into logic on an audio channel it changes this new file into a quieter slightly muffled version. however dragging it into the sampler preserves the sound of the 32 bit file - I decided to draw a midi note from the sampler with this sound loaded and exported the entire project file at 24bit (blank project file with just this 3 second sound playing out)

 

the result is great and this new 24bit exported version sounds the same as when I play the 32bit original file through my Mac just with the audio player... but dragging it into logic just creates this quieter version that seems lower in volume and quality

what is going on here? have I found good way of converting 32bit files without artefacts or mistakes? if I'm wrong is there a better option?

does sampler automatically convert 32bit files?

Edited by GuyBorlander
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Hm… the good ole "hi-res myth" in full effect, perhaps…? :)

From the Logic Pro user guide:

Quote

Sample Storage pop-up menu: Determines how Sampler handles the bit depth of samples. Choose one of the following:

  • Original: Loads samples into RAM at their original bit depth. These are converted to the internal 32-bit floating point format of the host application on playback.
  • 32-bit float: Samples are stored and loaded as 32-bit float files, which removes the need for real-time format conversions. Sampler handles 32-bit float samples more efficiently and can play back more voices simultaneously.
    Note: This requires twice as much RAM for 16-bit samples and a third more RAM for 24-bit samples.

If you know what you're doing, then 32-bit audio may make sense in certain scenarios and workflows.

Other than that, for a start on this complex topic, I highly recommend this entertaining reading:
https://people.xiph.org/~xiphmont/demo/neil-young.html
There are also a couple of corresponding and equally entertaining educational videos on the web available.

Edited by loukash
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I’m simply loading a single 2 second audio file into sampler that is 32bit. It sounds correct when played through sampler than when played back after dragging into Logic. The original file is about +4db so I have to turn it down on sampler by 4db and then it sounds just right without any distortion.

 

If I drag into Logic and play back Logic audio adjusts and takes away a lot of the power of the signal and squashes it and doesn’t just leave it alone. That’s what I’m saying 

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I’m simply loading a single 2 second audio file into sampler that is 32bit. It sounds correct when played through sampler than when played back after dragging into Logic. The original file is about +4db so I have to turn it down on sampler by 4db and then it sounds just right without any distortion.

If I drag into Logic and play back Logic automatically adjusts / deducts and takes away a lot of the power of the signal and squashes it and doesn’t just leave it alone. That’s what I’m saying. 

Edited by GuyBorlander
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22 minutes ago, GuyBorlander said:

here it is

The file has a true peak level of a whopping +4.05 dBFS. While it's possible to maintain this and work with it in a constant 32-bit floating audio workflow, plain converting to 24-bit or 16-bit would simply shave off the peaks down to 0.0 dBFS, resulting in a massively distorted signal.

Thus, if what you hear in Logic is the sound at reduced volume, then Logic is likely automatically performing the right thing and normalizing the level of the waveform so that the true peaks don't exceed 0.0 dBFS. This is also what you should be doing manually if converting floating 32-bit audio to 24-bit or lower.

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"Logic is likely automatically performing the right thing and normalizing the level of the waveform so that the true peaks don't exceed 0.0 dBF

so why does sampler using original setting not do this? Id prefer to load the original file in its entirety and then manually deduct myself rather than relying on Logics normalisation algorithm which is what it seems I am doing using sampler..

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sorry I still don't understand this I'm more of a producer than engineer or anything else

"Loads samples into RAM at their original bit depth. These are converted to the internal 32-bit floating point format of the host application on playback"

so what happens when I've drawn some notes in midi playing back this sound loaded in sampler and then exported it as a stem into 24bits.. has our 32bit file been downscaled to 24bits creating artefacts etc

I can only export and send stems out as 24bits you see.. because it is the standard and engineers don't like to work with 32bit float

"hi res myth"? not sure what this is

thank you for your replies.

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