AAC encoding from dff files Options

[hlloyge]

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.

[hlloyge]

Foobar can scan the file, but can't save the results into tags, because, AFAIK, dff has no tagging support. I've thought about that, it would be the best solution. And I don't know how to apply RG values while encoding - I could lower the output for file without RG info while playing back, I know to do that.
And although it plays fine in foobar2000 (AFAIK, foobar internally handles 32bit pcm so it will not clip), I will test out today afternoon how it sounds in iPod Touch.

Alternatively, I can transcode it to some lossless format which can handle 32bit wave input, tag it, and encode to aac.

This post has been edited by hlloyge: Feb 1 2012, 11:05

[nu774]

Alternatively, as long as the encoder can take it and encode it properly, you can adjust global gain value of resulting AAC afterwards with aacgain (LC-AAC only, though).