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AAC encoding from dff files
hlloyge
post Jan 31 2012, 10:51
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Hello all.

I have a question; I have a tune in dff (sacd rip) format which I obtained by shady sources, and it is not relevant for this discussion.

What I want to know is, how to encode it to aac? I am using foobar2000, and I loaded dff decoder plugin, and the song plays fine. But drum at the beginning peaks far above 0 dB, I guess it goes to +6. When I encode it to AAC, I use sox resampler plugin to convert it to 44100, but then it also peaks far above, but a bit less, I guess something about +3 or +4. I think it is because it is decoded AFAIK to 32bit float, which can handle higher peaks. I don't hear distortion whatsoever in original and encoded file, at least not on my desktop speakers, I haven't done any ABX test.

The question is - is this OK? I know mp3 doesn't have bit depth in the normal way wav file has; but I don't know about AAC. Can it handle that high input for encoding and decode it properly, without artifacts, or would I have to use some sort of peak limiting before encoding, or just decode it to wav and then load it up in Audacity and normalize it's peak limits to zero?

Thank you.
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saratoga
post Jan 31 2012, 21:20
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Can foobar replaygain scan the file? If so, the easiest/safest solution would be to replaygain scan it, then check the "prevent clipping" option when you convert it to AAC. This way it'll get scaled to 0dB before conversion.

This post has been edited by db1989: Jan 31 2012, 22:33
Reason for edit: removing unnecessary full quote of first post
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C.R.Helmrich
post Jan 31 2012, 22:49
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I agree with nu774 and saratoga. Given an encoder supporting floating-point input, a few sporadic peaks slightly above 0 dBFS in the input are OK, I guess. But remember that not every decoder can output floating-point PCM but rather truncates (and clips) to 16- or 24-bit before you can apply any level adjustments, and given such decoders you will risk audible clipping with above-0-dBFS encodings.

Chris


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