[Libav-user] Audio quality loss while encoding
nicolas.george at normalesup.org
Fri May 3 12:32:19 CEST 2013
Le duodi 12 floréal, an CCXXI, Claudio Freire a écrit :
> Ok... I had to forfeit my right to sleep, but here it is. Managed to
> make both the fast and 2loop methods work reasonably well (albeit I
> don't know why anyone would use fast, because it's not faster. I
> assume it's an experiment?).
> I don't have access to AAC standards, though, since they're paywalled
> (at least the ones I found), so most of the changes were done blind,
> assuming from what I know of audio codecs, and by trial and error.
> In particular, fast has some magic numbers there that I got from trial
> and error, and nothing more... profound.
> In any case, attached it is, git diff.
Thanks for the patch. I believe you should submit it to ffmpeg-devel, as it
will receive more attention.
Also, you should use git format-patch rather than git diff, so that author
information is included in the patch, and so would be the commit message,
where you can explain the changes for future reference.
A small style comment; I know nothing about AAC so I can not tell about the
validity of changes.
> + float lowlambda = av_clip(120.f / lambda, 0.25f, 1.f);
> + float rlambda = av_clip(120.f / lambda, 0.3f, 2.f);
> + const int minq = av_clip(2 * log2f(120.f / lambda) + 150, 100, 218 - SCALE_MAX_DIFF);
> + const int maxq = minq + SCALE_MAX_DIFF - 1;
Do you have a specific reason to use float instead of double?
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