[FFmpeg-devel] avfilter: add acontrast filter

Submitted by Paul B Mahol on Nov. 18, 2017, 10:44 a.m.

Details

Message ID 20171118104410.1508-1-onemda@gmail.com
State New
Headers show

Commit Message

Paul B Mahol Nov. 18, 2017, 10:44 a.m.
Signed-off-by: Paul B Mahol <onemda@gmail.com>
---
 doc/filters.texi           |  10 +++
 libavfilter/Makefile       |   1 +
 libavfilter/af_acontrast.c | 219 +++++++++++++++++++++++++++++++++++++++++++++
 libavfilter/allfilters.c   |   1 +
 4 files changed, 231 insertions(+)
 create mode 100644 libavfilter/af_acontrast.c

Comments

Rostislav Pehlivanov Nov. 18, 2017, 3:54 p.m.
On 18 November 2017 at 10:44, Paul B Mahol <onemda@gmail.com> wrote:

> Signed-off-by: Paul B Mahol <onemda@gmail.com>
> ---
>  doc/filters.texi           |  10 +++
>  libavfilter/Makefile       |   1 +
>  libavfilter/af_acontrast.c | 219 ++++++++++++++++++++++++++++++
> +++++++++++++++
>  libavfilter/allfilters.c   |   1 +
>  4 files changed, 231 insertions(+)
>  create mode 100644 libavfilter/af_acontrast.c
>
> diff --git a/doc/filters.texi b/doc/filters.texi
> index 5d99437871..e35952510b 100644
> --- a/doc/filters.texi
> +++ b/doc/filters.texi
> @@ -429,6 +429,16 @@ How much to use compressed signal in output. Default
> is 1.
>  Range is between 0 and 1.
>  @end table
>
> +@section acontrast
> +Simple audio dynamic range commpression/expansion filter.
> +
> +The filter accepts the following options:
> +
> +@table @option
> +@item c
> +Set contrast. Default is 33. Allowed range is between 0 and 100.
> +@end table
> +
>  @section acopy
>
>  Copy the input audio source unchanged to the output. This is mainly
> useful for
> diff --git a/libavfilter/Makefile b/libavfilter/Makefile
> index 9acae3ff5b..71c6333a52 100644
> --- a/libavfilter/Makefile
> +++ b/libavfilter/Makefile
> @@ -31,6 +31,7 @@ OBJS-$(CONFIG_QSVVPP)                        += qsvvpp.o
>  # audio filters
>  OBJS-$(CONFIG_ABENCH_FILTER)                 += f_bench.o
>  OBJS-$(CONFIG_ACOMPRESSOR_FILTER)            += af_sidechaincompress.o
> +OBJS-$(CONFIG_ACONTRAST_FILTER)              += af_acontrast.o
>  OBJS-$(CONFIG_ACOPY_FILTER)                  += af_acopy.o
>  OBJS-$(CONFIG_ACROSSFADE_FILTER)             += af_afade.o
>  OBJS-$(CONFIG_ACRUSHER_FILTER)               += af_acrusher.o
> diff --git a/libavfilter/af_acontrast.c b/libavfilter/af_acontrast.c
> new file mode 100644
> index 0000000000..38de08ffe5
> --- /dev/null
> +++ b/libavfilter/af_acontrast.c
> @@ -0,0 +1,219 @@
> +/*
> + * Copyright (c) 2008 Rob Sykes
> + * Copyright (c) 2017 Paul B Mahol
> + *
> + * 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 "libavutil/channel_layout.h"
> +#include "libavutil/opt.h"
> +#include "avfilter.h"
> +#include "audio.h"
> +#include "formats.h"
> +
> +typedef struct AudioContrastContext {
> +    const AVClass *class;
> +    float contrast;
> +    void (*filter)(void **dst, const void **src,
> +                   int nb_samples, int channels, float contrast);
> +} AudioContrastContext;
> +
> +#define OFFSET(x) offsetof(AudioContrastContext, x)
> +#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
> +
> +static const AVOption acontrast_options[] = {
> +    { "c", "set contrast", OFFSET(contrast), AV_OPT_TYPE_FLOAT,
> {.dbl=33}, 0, 100, A },
>

"contrast" instead of "c"? Not sure if single letter options are a good
idea.



> +    { NULL }
> +};
> +
> +AVFILTER_DEFINE_CLASS(acontrast);
> +
> +static int query_formats(AVFilterContext *ctx)
> +{
> +    AVFilterFormats *formats = NULL;
> +    AVFilterChannelLayouts *layouts = NULL;
> +    static const enum AVSampleFormat sample_fmts[] = {
> +        AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP,
> +        AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBLP,
> +        AV_SAMPLE_FMT_NONE
> +    };
> +    int ret;
> +
> +    formats = ff_make_format_list(sample_fmts);
> +    if (!formats)
> +        return AVERROR(ENOMEM);
> +    ret = ff_set_common_formats(ctx, formats);
> +    if (ret < 0)
> +        return ret;
> +
> +    layouts = ff_all_channel_counts();
> +    if (!layouts)
> +        return AVERROR(ENOMEM);
> +
> +    ret = ff_set_common_channel_layouts(ctx, layouts);
> +    if (ret < 0)
> +        return ret;
> +
> +    formats = ff_all_samplerates();
> +    return ff_set_common_samplerates(ctx, formats);
> +}
> +
> +static void filter_flt(void **d, const void **s,
> +                       int nb_samples, int channels,
> +                       float contrast)
> +{
> +    const float *src = s[0];
> +    float *dst = d[0];
> +    int n, c;
> +
> +    for (n = 0; n < nb_samples; n++) {
> +        for (c = 0; c < channels; c++) {
> +            double d = src[c] * M_PI_2;
> +
> +            dst[c] = sin(d + contrast * sin(d * 4));
>

sinf() instead of sin()



> +        }
> +
> +        dst += c;
> +        src += c;
> +    }
> +}
> +
> +static void filter_dbl(void **d, const void **s,
> +                       int nb_samples, int channels,
> +                       float contrast)
> +{
> +    const double *src = s[0];
> +    double *dst = d[0];
> +    int n, c;
> +
> +    for (n = 0; n < nb_samples; n++) {
> +        for (c = 0; c < channels; c++) {
> +            double d = src[c] * M_PI_2;
> +
> +            dst[c] = sin(d + contrast * sin(d * 4));
> +        }
> +
> +        dst += c;
> +        src += c;
> +    }
> +}
> +
> +static void filter_fltp(void **d, const void **s,
> +                        int nb_samples, int channels,
> +                        float contrast)
> +{
> +    int n, c;
> +
> +    for (c = 0; c < channels; c++) {
> +        const float *src = s[c];
> +        float *dst = d[c];
> +
> +        for (n = 0; n < nb_samples; n++) {
> +            double d = src[n] * M_PI_2;
> +
> +            dst[n] = sin(d + contrast * sin(d * 4));
>

sinf() instead of sin()



> +        }
> +    }
> +}
> +
> +static void filter_dblp(void **d, const void **s,
> +                        int nb_samples, int channels,
> +                        float contrast)
> +{
> +    int n, c;
> +
> +    for (c = 0; c < channels; c++) {
> +        const double *src = s[c];
> +        double *dst = d[c];
> +
> +        for (n = 0; n < nb_samples; n++) {
> +            double d = src[n] * M_PI_2;
> +
> +            dst[n] = sin(d + contrast * sin(d * 4));
> +        }
> +    }
> +}
>

Could you do the filtering in-place? Via av_frame_make_writeable?



> +
> +static int config_input(AVFilterLink *inlink)
> +{
> +    AVFilterContext *ctx = inlink->dst;
> +    AudioContrastContext *s    = ctx->priv;
> +
> +    switch (inlink->format) {
> +    case AV_SAMPLE_FMT_FLT:  s->filter = filter_flt;  break;
> +    case AV_SAMPLE_FMT_DBL:  s->filter = filter_dbl;  break;
> +    case AV_SAMPLE_FMT_FLTP: s->filter = filter_fltp; break;
> +    case AV_SAMPLE_FMT_DBLP: s->filter = filter_dblp; break;
> +    }
> +
> +    return 0;
> +}
> +
> +static int filter_frame(AVFilterLink *inlink, AVFrame *in)
> +{
> +    AVFilterContext *ctx = inlink->dst;
> +    AVFilterLink *outlink = ctx->outputs[0];
> +    AudioContrastContext *s = ctx->priv;
> +    AVFrame *out;
> +
> +    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);
> +    }
> +
> +    s->filter((void **)out->extended_data, (const void
> **)in->extended_data,
> +              in->nb_samples, in->channels, s->contrast / 750);
>

Divide s->contrast by 750 during init?


> +
> +    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_acontrast = {
> +    .name           = "acontrast",
> +    .description    = NULL_IF_CONFIG_SMALL("Simple audio dynamic range
> compression/expansion filter."),
> +    .query_formats  = query_formats,
> +    .priv_size      = sizeof(AudioContrastContext),
> +    .priv_class     = &acontrast_class,
> +    .inputs         = inputs,
> +    .outputs        = outputs,
> +};
> diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
> index a838309569..6d92b3ab5a 100644
> --- a/libavfilter/allfilters.c
> +++ b/libavfilter/allfilters.c
> @@ -42,6 +42,7 @@ static void register_all(void)
>  {
>      REGISTER_FILTER(ABENCH,         abench,         af);
>      REGISTER_FILTER(ACOMPRESSOR,    acompressor,    af);
> +    REGISTER_FILTER(ACONTRAST,      acontrast,      af);
>      REGISTER_FILTER(ACOPY,          acopy,          af);
>      REGISTER_FILTER(ACROSSFADE,     acrossfade,     af);
>      REGISTER_FILTER(ACRUSHER,       acrusher,       af);
> --
> 2.11.0
>
> _______________________________________________
> ffmpeg-devel mailing list
> ffmpeg-devel@ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-devel
>


Apart from that lgtm
Paul B Mahol Nov. 18, 2017, 4:03 p.m.
On 11/18/17, Rostislav Pehlivanov <atomnuker@gmail.com> wrote:
> On 18 November 2017 at 10:44, Paul B Mahol <onemda@gmail.com> wrote:
>
>> Signed-off-by: Paul B Mahol <onemda@gmail.com>
>> ---
>>  doc/filters.texi           |  10 +++
>>  libavfilter/Makefile       |   1 +
>>  libavfilter/af_acontrast.c | 219 ++++++++++++++++++++++++++++++
>> +++++++++++++++
>>  libavfilter/allfilters.c   |   1 +
>>  4 files changed, 231 insertions(+)
>>  create mode 100644 libavfilter/af_acontrast.c
>>
>> diff --git a/doc/filters.texi b/doc/filters.texi
>> index 5d99437871..e35952510b 100644
>> --- a/doc/filters.texi
>> +++ b/doc/filters.texi
>> @@ -429,6 +429,16 @@ How much to use compressed signal in output. Default
>> is 1.
>>  Range is between 0 and 1.
>>  @end table
>>
>> +@section acontrast
>> +Simple audio dynamic range commpression/expansion filter.
>> +
>> +The filter accepts the following options:
>> +
>> +@table @option
>> +@item c
>> +Set contrast. Default is 33. Allowed range is between 0 and 100.
>> +@end table
>> +
>>  @section acopy
>>
>>  Copy the input audio source unchanged to the output. This is mainly
>> useful for
>> diff --git a/libavfilter/Makefile b/libavfilter/Makefile
>> index 9acae3ff5b..71c6333a52 100644
>> --- a/libavfilter/Makefile
>> +++ b/libavfilter/Makefile
>> @@ -31,6 +31,7 @@ OBJS-$(CONFIG_QSVVPP)                        += qsvvpp.o
>>  # audio filters
>>  OBJS-$(CONFIG_ABENCH_FILTER)                 += f_bench.o
>>  OBJS-$(CONFIG_ACOMPRESSOR_FILTER)            += af_sidechaincompress.o
>> +OBJS-$(CONFIG_ACONTRAST_FILTER)              += af_acontrast.o
>>  OBJS-$(CONFIG_ACOPY_FILTER)                  += af_acopy.o
>>  OBJS-$(CONFIG_ACROSSFADE_FILTER)             += af_afade.o
>>  OBJS-$(CONFIG_ACRUSHER_FILTER)               += af_acrusher.o
>> diff --git a/libavfilter/af_acontrast.c b/libavfilter/af_acontrast.c
>> new file mode 100644
>> index 0000000000..38de08ffe5
>> --- /dev/null
>> +++ b/libavfilter/af_acontrast.c
>> @@ -0,0 +1,219 @@
>> +/*
>> + * Copyright (c) 2008 Rob Sykes
>> + * Copyright (c) 2017 Paul B Mahol
>> + *
>> + * 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 "libavutil/channel_layout.h"
>> +#include "libavutil/opt.h"
>> +#include "avfilter.h"
>> +#include "audio.h"
>> +#include "formats.h"
>> +
>> +typedef struct AudioContrastContext {
>> +    const AVClass *class;
>> +    float contrast;
>> +    void (*filter)(void **dst, const void **src,
>> +                   int nb_samples, int channels, float contrast);
>> +} AudioContrastContext;
>> +
>> +#define OFFSET(x) offsetof(AudioContrastContext, x)
>> +#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
>> +
>> +static const AVOption acontrast_options[] = {
>> +    { "c", "set contrast", OFFSET(contrast), AV_OPT_TYPE_FLOAT,
>> {.dbl=33}, 0, 100, A },
>>
>
> "contrast" instead of "c"? Not sure if single letter options are a good
> idea.
>
>
>
>> +    { NULL }
>> +};
>> +
>> +AVFILTER_DEFINE_CLASS(acontrast);
>> +
>> +static int query_formats(AVFilterContext *ctx)
>> +{
>> +    AVFilterFormats *formats = NULL;
>> +    AVFilterChannelLayouts *layouts = NULL;
>> +    static const enum AVSampleFormat sample_fmts[] = {
>> +        AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP,
>> +        AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBLP,
>> +        AV_SAMPLE_FMT_NONE
>> +    };
>> +    int ret;
>> +
>> +    formats = ff_make_format_list(sample_fmts);
>> +    if (!formats)
>> +        return AVERROR(ENOMEM);
>> +    ret = ff_set_common_formats(ctx, formats);
>> +    if (ret < 0)
>> +        return ret;
>> +
>> +    layouts = ff_all_channel_counts();
>> +    if (!layouts)
>> +        return AVERROR(ENOMEM);
>> +
>> +    ret = ff_set_common_channel_layouts(ctx, layouts);
>> +    if (ret < 0)
>> +        return ret;
>> +
>> +    formats = ff_all_samplerates();
>> +    return ff_set_common_samplerates(ctx, formats);
>> +}
>> +
>> +static void filter_flt(void **d, const void **s,
>> +                       int nb_samples, int channels,
>> +                       float contrast)
>> +{
>> +    const float *src = s[0];
>> +    float *dst = d[0];
>> +    int n, c;
>> +
>> +    for (n = 0; n < nb_samples; n++) {
>> +        for (c = 0; c < channels; c++) {
>> +            double d = src[c] * M_PI_2;
>> +
>> +            dst[c] = sin(d + contrast * sin(d * 4));
>>
>
> sinf() instead of sin()

ok

>
>
>
>> +        }
>> +
>> +        dst += c;
>> +        src += c;
>> +    }
>> +}
>> +
>> +static void filter_dbl(void **d, const void **s,
>> +                       int nb_samples, int channels,
>> +                       float contrast)
>> +{
>> +    const double *src = s[0];
>> +    double *dst = d[0];
>> +    int n, c;
>> +
>> +    for (n = 0; n < nb_samples; n++) {
>> +        for (c = 0; c < channels; c++) {
>> +            double d = src[c] * M_PI_2;
>> +
>> +            dst[c] = sin(d + contrast * sin(d * 4));
>> +        }
>> +
>> +        dst += c;
>> +        src += c;
>> +    }
>> +}
>> +
>> +static void filter_fltp(void **d, const void **s,
>> +                        int nb_samples, int channels,
>> +                        float contrast)
>> +{
>> +    int n, c;
>> +
>> +    for (c = 0; c < channels; c++) {
>> +        const float *src = s[c];
>> +        float *dst = d[c];
>> +
>> +        for (n = 0; n < nb_samples; n++) {
>> +            double d = src[n] * M_PI_2;
>> +
>> +            dst[n] = sin(d + contrast * sin(d * 4));
>>
>
> sinf() instead of sin()
>

ok

>
>
>> +        }
>> +    }
>> +}
>> +
>> +static void filter_dblp(void **d, const void **s,
>> +                        int nb_samples, int channels,
>> +                        float contrast)
>> +{
>> +    int n, c;
>> +
>> +    for (c = 0; c < channels; c++) {
>> +        const double *src = s[c];
>> +        double *dst = d[c];
>> +
>> +        for (n = 0; n < nb_samples; n++) {
>> +            double d = src[n] * M_PI_2;
>> +
>> +            dst[n] = sin(d + contrast * sin(d * 4));
>> +        }
>> +    }
>> +}
>>
>
> Could you do the filtering in-place? Via av_frame_make_writeable?
>

Both are supported.

>
>
>> +
>> +static int config_input(AVFilterLink *inlink)
>> +{
>> +    AVFilterContext *ctx = inlink->dst;
>> +    AudioContrastContext *s    = ctx->priv;
>> +
>> +    switch (inlink->format) {
>> +    case AV_SAMPLE_FMT_FLT:  s->filter = filter_flt;  break;
>> +    case AV_SAMPLE_FMT_DBL:  s->filter = filter_dbl;  break;
>> +    case AV_SAMPLE_FMT_FLTP: s->filter = filter_fltp; break;
>> +    case AV_SAMPLE_FMT_DBLP: s->filter = filter_dblp; break;
>> +    }
>> +
>> +    return 0;
>> +}
>> +
>> +static int filter_frame(AVFilterLink *inlink, AVFrame *in)
>> +{
>> +    AVFilterContext *ctx = inlink->dst;
>> +    AVFilterLink *outlink = ctx->outputs[0];
>> +    AudioContrastContext *s = ctx->priv;
>> +    AVFrame *out;
>> +
>> +    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);
>> +    }
>> +
>> +    s->filter((void **)out->extended_data, (const void
>> **)in->extended_data,
>> +              in->nb_samples, in->channels, s->contrast / 750);
>>
>
> Divide s->contrast by 750 during init?

Doesn't cost much, and also it is less lines this way.

Patch hide | download patch | download mbox

diff --git a/doc/filters.texi b/doc/filters.texi
index 5d99437871..e35952510b 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -429,6 +429,16 @@  How much to use compressed signal in output. Default is 1.
 Range is between 0 and 1.
 @end table
 
+@section acontrast
+Simple audio dynamic range commpression/expansion filter.
+
+The filter accepts the following options:
+
+@table @option
+@item c
+Set contrast. Default is 33. Allowed range is between 0 and 100.
+@end table
+
 @section acopy
 
 Copy the input audio source unchanged to the output. This is mainly useful for
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 9acae3ff5b..71c6333a52 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -31,6 +31,7 @@  OBJS-$(CONFIG_QSVVPP)                        += qsvvpp.o
 # audio filters
 OBJS-$(CONFIG_ABENCH_FILTER)                 += f_bench.o
 OBJS-$(CONFIG_ACOMPRESSOR_FILTER)            += af_sidechaincompress.o
+OBJS-$(CONFIG_ACONTRAST_FILTER)              += af_acontrast.o
 OBJS-$(CONFIG_ACOPY_FILTER)                  += af_acopy.o
 OBJS-$(CONFIG_ACROSSFADE_FILTER)             += af_afade.o
 OBJS-$(CONFIG_ACRUSHER_FILTER)               += af_acrusher.o
diff --git a/libavfilter/af_acontrast.c b/libavfilter/af_acontrast.c
new file mode 100644
index 0000000000..38de08ffe5
--- /dev/null
+++ b/libavfilter/af_acontrast.c
@@ -0,0 +1,219 @@ 
+/*
+ * Copyright (c) 2008 Rob Sykes
+ * Copyright (c) 2017 Paul B Mahol
+ *
+ * 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 "libavutil/channel_layout.h"
+#include "libavutil/opt.h"
+#include "avfilter.h"
+#include "audio.h"
+#include "formats.h"
+
+typedef struct AudioContrastContext {
+    const AVClass *class;
+    float contrast;
+    void (*filter)(void **dst, const void **src,
+                   int nb_samples, int channels, float contrast);
+} AudioContrastContext;
+
+#define OFFSET(x) offsetof(AudioContrastContext, x)
+#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
+
+static const AVOption acontrast_options[] = {
+    { "c", "set contrast", OFFSET(contrast), AV_OPT_TYPE_FLOAT, {.dbl=33}, 0, 100, A },
+    { NULL }
+};
+
+AVFILTER_DEFINE_CLASS(acontrast);
+
+static int query_formats(AVFilterContext *ctx)
+{
+    AVFilterFormats *formats = NULL;
+    AVFilterChannelLayouts *layouts = NULL;
+    static const enum AVSampleFormat sample_fmts[] = {
+        AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP,
+        AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBLP,
+        AV_SAMPLE_FMT_NONE
+    };
+    int ret;
+
+    formats = ff_make_format_list(sample_fmts);
+    if (!formats)
+        return AVERROR(ENOMEM);
+    ret = ff_set_common_formats(ctx, formats);
+    if (ret < 0)
+        return ret;
+
+    layouts = ff_all_channel_counts();
+    if (!layouts)
+        return AVERROR(ENOMEM);
+
+    ret = ff_set_common_channel_layouts(ctx, layouts);
+    if (ret < 0)
+        return ret;
+
+    formats = ff_all_samplerates();
+    return ff_set_common_samplerates(ctx, formats);
+}
+
+static void filter_flt(void **d, const void **s,
+                       int nb_samples, int channels,
+                       float contrast)
+{
+    const float *src = s[0];
+    float *dst = d[0];
+    int n, c;
+
+    for (n = 0; n < nb_samples; n++) {
+        for (c = 0; c < channels; c++) {
+            double d = src[c] * M_PI_2;
+
+            dst[c] = sin(d + contrast * sin(d * 4));
+        }
+
+        dst += c;
+        src += c;
+    }
+}
+
+static void filter_dbl(void **d, const void **s,
+                       int nb_samples, int channels,
+                       float contrast)
+{
+    const double *src = s[0];
+    double *dst = d[0];
+    int n, c;
+
+    for (n = 0; n < nb_samples; n++) {
+        for (c = 0; c < channels; c++) {
+            double d = src[c] * M_PI_2;
+
+            dst[c] = sin(d + contrast * sin(d * 4));
+        }
+
+        dst += c;
+        src += c;
+    }
+}
+
+static void filter_fltp(void **d, const void **s,
+                        int nb_samples, int channels,
+                        float contrast)
+{
+    int n, c;
+
+    for (c = 0; c < channels; c++) {
+        const float *src = s[c];
+        float *dst = d[c];
+
+        for (n = 0; n < nb_samples; n++) {
+            double d = src[n] * M_PI_2;
+
+            dst[n] = sin(d + contrast * sin(d * 4));
+        }
+    }
+}
+
+static void filter_dblp(void **d, const void **s,
+                        int nb_samples, int channels,
+                        float contrast)
+{
+    int n, c;
+
+    for (c = 0; c < channels; c++) {
+        const double *src = s[c];
+        double *dst = d[c];
+
+        for (n = 0; n < nb_samples; n++) {
+            double d = src[n] * M_PI_2;
+
+            dst[n] = sin(d + contrast * sin(d * 4));
+        }
+    }
+}
+
+static int config_input(AVFilterLink *inlink)
+{
+    AVFilterContext *ctx = inlink->dst;
+    AudioContrastContext *s    = ctx->priv;
+
+    switch (inlink->format) {
+    case AV_SAMPLE_FMT_FLT:  s->filter = filter_flt;  break;
+    case AV_SAMPLE_FMT_DBL:  s->filter = filter_dbl;  break;
+    case AV_SAMPLE_FMT_FLTP: s->filter = filter_fltp; break;
+    case AV_SAMPLE_FMT_DBLP: s->filter = filter_dblp; break;
+    }
+
+    return 0;
+}
+
+static int filter_frame(AVFilterLink *inlink, AVFrame *in)
+{
+    AVFilterContext *ctx = inlink->dst;
+    AVFilterLink *outlink = ctx->outputs[0];
+    AudioContrastContext *s = ctx->priv;
+    AVFrame *out;
+
+    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);
+    }
+
+    s->filter((void **)out->extended_data, (const void **)in->extended_data,
+              in->nb_samples, in->channels, s->contrast / 750);
+
+    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_acontrast = {
+    .name           = "acontrast",
+    .description    = NULL_IF_CONFIG_SMALL("Simple audio dynamic range compression/expansion filter."),
+    .query_formats  = query_formats,
+    .priv_size      = sizeof(AudioContrastContext),
+    .priv_class     = &acontrast_class,
+    .inputs         = inputs,
+    .outputs        = outputs,
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index a838309569..6d92b3ab5a 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -42,6 +42,7 @@  static void register_all(void)
 {
     REGISTER_FILTER(ABENCH,         abench,         af);
     REGISTER_FILTER(ACOMPRESSOR,    acompressor,    af);
+    REGISTER_FILTER(ACONTRAST,      acontrast,      af);
     REGISTER_FILTER(ACOPY,          acopy,          af);
     REGISTER_FILTER(ACROSSFADE,     acrossfade,     af);
     REGISTER_FILTER(ACRUSHER,       acrusher,       af);