[FFmpeg-devel] [PATCH] examples: add resampling_audio.c file

[Stefano Sabatini]

On date Monday 2012-12-03 16:42:03 +0100, Michael Niedermayer encoded:
> On Mon, Dec 03, 2012 at 01:31:34PM +0100, Stefano Sabatini wrote:
> > On date Sunday 2012-12-02 00:34:03 +0100, Michael Niedermayer encoded:
> > > On Sat, Dec 01, 2012 at 08:21:47PM +0100, Stefano Sabatini wrote:
[...]
> > [...] 
> > > > +    /* allocate source and destination samples buffers */
> > > > +
> > > > +    src_nb_channels = av_get_channel_layout_nb_channels(src_ch_layout);
> > > > +    ret = alloc_samples_array_and_data(&src_data, &src_linesize, src_nb_channels,
> > > > +                                       src_nb_samples, src_sample_fmt, 0);
> > > > +    if (ret < 0) {
> > > > +        fprintf(stderr, "Could not allocate source samples\n");
> > > > +        goto end;
> > > > +    }
> > > > +
> > > 
> > 
> > > > +    /* compute the number of converted samples: buffering is avoided
> > > > +     * ensuring that the output buffer will contain at least all the
> > > > +     * converted input samples */
> > > > +    dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, dst_rate) + src_nb_samples,
> > > > +                                    dst_rate, src_rate, AV_ROUND_UP);
> > > 
> > > isnt this mixing up src and dst rates ?
> > 
> > Yes, it is: 
> > 
> > N2 = N1 * R2 / R1
> > 
> > and it is copied from the swresample.h doxy.
> 
> from where?
> 
> the doxy says this:
> 
>  * 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

And I'm not sure why swr_get_delay() is useful/required.
-- 
FFmpeg = Frightening and Funny Murdering Perfectionist Elected Gadget