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

Sanchit Sinha sanchit15083 at iiitd.ac.in
Fri Mar 10 22:02:30 EET 2017


Okay made some changes,

>From c0c1a1e7d4ad1fcbd96827725a47af20145c7621 Mon Sep 17 00:00:00 2001
From: Sanchit Sinha <sanchit15083 at iiitd.ac.in>
Date: Sat, 11 Mar 2017 01:27:38 +0530
Subject: [PATCH] Changes to af_ambisonic.c, makefile and allfilters.c

---
 libavfilter/af_ambisonic.c | 136
+++++++++++++++++++++++++++++++++++++++++++++
 1 file changed, 136 insertions(+)
 create mode 100644 libavfilter/af_ambisonic.c

diff --git a/libavfilter/af_ambisonic.c b/libavfilter/af_ambisonic.c
new file mode 100644
index 0000000..03409a6
--- /dev/null
+++ b/libavfilter/af_ambisonic.c
@@ -0,0 +1,136 @@
+/*
+ * 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"
+#include <math.h>
+
+typedef struct AmbisonicContext {
+ const AVClass *class;
+ /*Not needed, if any new variables are to be used, this struct can be
populated(f)*/
+
+} AmbisonicContext;
+
+#define OFFSET(x) offsetof(AmbisonicContext, x)
+
+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;
+ 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];
+
+ float root8= sqrt(8);
+ float lf=0,lb=0,rb=0,rf=0;
+ for(itr=0;itr<in->nb_samples;itr++)
+ {
+ /*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);
+
+ lf = root8 * (2*(*w)+*x+*y);
+ lb = root8 * (2*(*w)-*x+*y);
+ rb = root8 * (2*(*w)-*x-*y);
+ rf = root8 * (2*(*w)+*x-*y);
+
+ 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
-- 
2.7.4





On Sat, Mar 11, 2017 at 12:33 AM, Paul B Mahol <onemda at gmail.com> wrote:

> 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.
> _______________________________________________
> ffmpeg-devel mailing list
> ffmpeg-devel at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-devel
>



-- 
Sanchit Sinha
B.Tech- CSE
IIIT-Delhi
Roll-2015083


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