[FFmpeg-devel] [PATCH] examples: add resampling_audio.c file
michaelni at gmx.at
Tue Dec 4 18:14:53 CET 2012
On Tue, Dec 04, 2012 at 10:58:48AM +0100, Stefano Sabatini wrote:
> On date Tuesday 2012-12-04 01:54:21 +0100, Michael Niedermayer encoded:
> > On Tue, Dec 04, 2012 at 12:12:53AM +0100, Stefano Sabatini wrote:
> > > > * av_opt_set_int(swr, "in_sample_rate", 48000, 0);
> > > > * av_opt_set_int(swr, "out_sample_rate", 44100, 0);
> > > > [....]
> > > > * int out_samples = av_rescale_rnd(swr_get_delay(swr, 48000) +
> > > > * in_samples, 44100, 48000, AV_ROUND_UP);
> > >
> > > dst_rate = 44100
> > > src_rate = 48000
> > this would make:
> > > > > > > + dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, dst_rate) + src_nb_samples,
> > > > > > > + dst_rate, src_rate, AV_ROUND_UP);
> > to this:
> > > > > > > + dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, 44100) + src_nb_samples,
> > > > > > > + 44100, 48000, AV_ROUND_UP);
> > which is not what the example says
> > >
> > > And I'm not sure why swr_get_delay() is useful/required.
> > If you ever feed "more" in than take out then these samples get
> > buffered. Such samples would be stuck in there until flushing
> > at the end without the swr_get_delay().
> I realized that the swr_get_delay() in the example is useless, since it
> is called before the first conversion is performed (so it is always
> 0). On the other hand the destination size is computed so that it
> should contain *all* the converted samples, so I assume that if the
> output buffer is big enough then no buffering will occurr at all. Is
> this correct?
resampling needs a delay and this delay may vary slightly due filter
parameters and timestamp drift.
this variation may occasionally cause a tiny bit more samples to be
available than the rounded up value.
Michael GnuPG fingerprint: 9FF2128B147EF6730BADF133611EC787040B0FAB
I have never wished to cater to the crowd; for what I know they do not
approve, and what they approve I do not know. -- Epicurus
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