[FFmpeg-user] Preserve bitdepth of input audio file during conversion to wav

Michal Šmucr msmucr at gmail.com
Wed Nov 27 14:51:07 CET 2013


Hello Carl Eugen,

thank you for reply.
I wrote that command to question only as an example, but if you'd like
to see it, i have attached output from one conversion.

I understand to behavior of FFmpeg and its basic logic with setting
default options of target format, which is guessed from output file
extension. But i think, it is not good in case of lossless audio
formats, i should wrote it clearly, that i aimed only to that. Main
point of its usage is, that these codecs don't modify PCM content
(regardless someone will hear that or not, that is different question
to me), so you can use it for instance during archiving. Its coding
and decoding should be transparent. And in this particular case -
conversion to uncompressed WAV (or AIFF), FFmpeg should (IMO) output
some default flavor of codec (eg. most used signed LE) and pick
shortest appropriate wordlength to avoid truncation.
For sure, it is not error, but rather usability thing. Generally,
using of FFmpeg for this tasks is very convenient, because it is not
necessary to utilize special tool for each lossless codec (eg. flac,
wvunpack, wavpack.. plus many which don't have its unix tool) and have
also useful mappings between particular metadata formats, but if you
try to use it with mixed bitdepths, it falls short without mentioned
wrapper. So i asked mainly, if i didn't miss some option for that.

Michal


2013/11/27 Carl Eugen Hoyos <cehoyos at ag.or.at>:
> Michal Šmucr <msmucr <at> gmail.com> writes:
>
>> When i use straight "ffmpeg -i whatever-24bit.flac output.wav",
>> resulting file is always coded as PCM_S16LE.
>
> (Complete, uncut console output missing.)
>
> If you do not specify a (audio) codec with -acodec xyz
> ffmpeg will use the one that is set as default for the
> file type you used. In your case you chose "wav" as
> file type and since you did not specify a codec (and
> since no human being can distinguish between the
> original 32bit source and the 16bit encoding), 16bit
> pcm is used.
>
> Carl Eugen
>
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> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
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