[FFmpeg-devel] [PATCH] libavfilter/af_ambisonic.c Added File for Ambisonic Filter

[Paul B Mahol]

On 3/10/17, Sanchit Sinha <sanchit15083 at iiitd.ac.in> wrote:
> libavfilter/af_ambisonic.c | 139
> +++++++++++++++++++++++++++++++++++++++++++++
>  1 file changed, 139 insertions(+)
>  create mode 100644 libavfilter/af_ambisonic.c
>

Incomplete patch.

> diff --git a/libavfilter/af_ambisonic.c b/libavfilter/af_ambisonic.c
> new file mode 100644
> index 0000000..98b0e44
> --- /dev/null
> +++ b/libavfilter/af_ambisonic.c
> @@ -0,0 +1,139 @@
> +/*
> + * This file is part of FFmpeg.
> + *
> + * FFmpeg is free software; you can redistribute it and/or
> + * modify it under the terms of the GNU Lesser General Public
> + * License as published by the Free Software Foundation; either
> + * version 2.1 of the License, or (at your option) any later version.
> + *
> + * FFmpeg is distributed in the hope that it will be useful,
> + * but WITHOUT ANY WARRANTY; without even the implied warranty of
> + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
> + * Lesser General Public License for more details.
> + *
> + * You should have received a copy of the GNU Lesser General Public
> + * License along with FFmpeg; if not, write to the Free Software
> + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA
> 02110-1301 USA
> + */
> +
> +#include <stdio.h>
> +#include "libavutil/avstring.h"
> +#include "libavutil/channel_layout.h"
> +#include "libavutil/opt.h"
> +#include "audio.h"
> +#include "avfilter.h"
> +#include "formats.h"
> +
> +#define root8 2.828

One can use sqrt(8) just fine.

> +
> +typedef struct AmbisonicContext {
> +    const AVClass *class;
> +    /*Not needed, if any new variables are to be used, this struct can be
> populated*/
> +
> +} AmbisonicContext;
> +
> +#define OFFSET(x) offsetof(AmbisonicContext, x) //Future use(maybe)
> +
> +static const AVOption ambisonic_options[] = {  //square will be an option
> +{NULL}
> +};
> +
> +AVFILTER_DEFINE_CLASS(ambisonic);
> +static int query_formats(AVFilterContext *ctx)
> +{
> +    AVFilterFormats *formats = NULL;
> +    AVFilterChannelLayouts *layout = NULL;
> +    int ret;
> +        if ((ret = ff_add_format             (&formats, AV_SAMPLE_FMT_FLTP
>   )) < 0 ||
> +        (ret = ff_set_common_formats         (ctx     , formats
>  )) < 0 ||
> +        (ret = ff_add_channel_layout         (&layout ,
> AV_CH_LAYOUT_4POINT0)) < 0 ||
> +        (ret = ff_set_common_channel_layouts (ctx     , layout
> )) < 0)
> +        return ret;
> +    formats = ff_all_samplerates();
> +    return ff_set_common_samplerates(ctx, formats);
> +}
> +static int filter_frame(AVFilterLink *inlink, AVFrame *in)
> +{
> +    AVFilterContext *ctx = inlink->dst;
> +    AVFilterLink *outlink = ctx->outputs[0];
> +    /*If variables used, this has to be created*/
> +    // AmbisonicContext *s = ctx->priv;
> +    // const float *src = (const float *)in->data[0];
> +    // float *dst;
> +    float *lf,*lb,*rb,*rf;
> +    AVFrame *out;
> +    int itr;
> +    float *w,*x,*y;
> +
> +    if (av_frame_is_writable(in)) {
> +        out = in;
> +    } else {
> +        out = ff_get_audio_buffer(inlink, in->nb_samples);
> +        if (!out) {
> +            av_frame_free(&in);
> +            return AVERROR(ENOMEM);
> +        }
> +        av_frame_copy_props(out, in);
> +    }
> +
> +    /*If planar samples are used, output must be put in dst*/
> +    //dst = (float *)out->data[0];
> +
> +    lf = (float *)malloc(sizeof(float));
> +    lb = (float *)malloc(sizeof(float));
> +    rb = (float *)malloc(sizeof(float));
> +    rf = (float *)malloc(sizeof(float));
> +

Why? This is unacceptable. Use normal float variables.

> +    for(itr=0;itr<in->nb_samples;itr++)
> +    {
> +        *lf=0,*lb=0,*rb=0,*rf=0;
> +        /*Float pointers to the samples*/
> +        w=(float *)(*(in->extended_data)+itr);
> +        x=(float *)(*(in->extended_data+1)+itr);
> +        y=(float *)(*(in->extended_data+2)+itr);

This can be simplified and moved above loop.
Good understanding of pointers is mandatory.

> +
> +        *lf = root8 * (2*(*w)+*x+*y);
> +        *lb = root8 * (2*(*w)-*x+*y);
> +        *rb = root8 * (2*(*w)-*x-*y);
> +        *rf = root8 * (2*(*w)+*x-*y);
> +
> +        /*Mathematical coefficients taken from :
> https://en.wikipedia.org/wiki/Ambisonics*/

Remove this comment.

> +        out->extended_data[0][itr]= *lf;
> +        out->extended_data[1][itr]= *lb;
> +        out->extended_data[2][itr]= *rb;
> +        out->extended_data[3][itr]= *rf;
> +    }
> +
> +    if (out != in)
> +        av_frame_free(&in);
> +    return ff_filter_frame(outlink, out);
> +}
> +
> +static const AVFilterPad inputs[] = {
> +    {
> +        .name         = "default",
> +        .type         = AVMEDIA_TYPE_AUDIO,
> +        .filter_frame = filter_frame,
> +        // .config_props = config_input,
> +    },
> +    { NULL }
> +};
> +
> +static const AVFilterPad outputs[] = {
> +    {
> +        .name = "default",
> +        .type = AVMEDIA_TYPE_AUDIO,
> +    },
> +    { NULL }
> +};
> +
> +AVFilter ff_af_ambisonic = {
> +    .name           = "ambisonic",
> +    .description    = NULL_IF_CONFIG_SMALL("An ambisonic filter"),
> +    .query_formats  = query_formats,
> +    .priv_size      = sizeof(AmbisonicContext),
> +    .priv_class     = &ambisonic_class,
> +    // .uninit         = uninit,
> +    .inputs         = inputs,
> +    .outputs        = outputs,
> +};
> \ No newline at end of file

Plese fix your editor or use something else less broken.
I hope you are not using MS notepad or MS Word.