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

[Stefano Sabatini]

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:
[...]
> > +static int get_format_from_sample_fmt(const char **fmt,
> > +                                      enum AVSampleFormat sample_fmt)
> > +{
> > +    int i;
> > +    struct sample_fmt_entry {
> > +        enum AVSampleFormat sample_fmt; const char *fmt_be, *fmt_le;
> > +    } sample_fmt_entries[] = {
> > +        { AV_SAMPLE_FMT_U8,  "u8",    "u8"    },
> > +        { AV_SAMPLE_FMT_S16, "s16be", "s16le" },
> > +        { AV_SAMPLE_FMT_S32, "s32be", "s32le" },
> > +        { AV_SAMPLE_FMT_FLT, "f32be", "f32le" },
> > +        { AV_SAMPLE_FMT_DBL, "f64be", "f64le" },
> > +    };
> > +    *fmt = NULL;
> > +
> > +    for (i = 0; i < FF_ARRAY_ELEMS(sample_fmt_entries); i++) {
> > +        struct sample_fmt_entry *entry = &sample_fmt_entries[i];
> > +        if (sample_fmt == entry->sample_fmt) {
> > +            *fmt = AV_NE(entry->fmt_be, entry->fmt_le);
> > +            return 0;
> > +        }
> > +    }
> 

> something like
> { AV_SAMPLE_FMT_U8,  AV_NE("u8",    "u8"   )    },
> { AV_SAMPLE_FMT_S16, AV_NE("s16be", "s16le")    },
> ...
> should be slightly simpler

Tried it, but I prefer slightly the current variant (more information
stored in the struct), and requires less characters.

[...] 
> > +    /* 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.
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