avfilter: add acontrast filter

Signed-off-by: Paul B Mahol <onemda@gmail.com>

Changelog

@@ -16,6 +16,7 @@ version <next>:
 - NVIDIA NVDEC-accelerated H.264, HEVC, MPEG-2, VC1 and VP9 hwaccel decoding
 - Intel QSV-accelerated overlay filter
 - mcompand audio filter
+- acontrast audio filter
 
 
 version 3.4:

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 contrast
+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

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

libavfilter/af_acontrast.c

+/*
+ * 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[] = {
+    { "contrast", "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++) {
+            float d = src[c] * M_PI_2;
+
+            dst[c] = sinf(d + contrast * sinf(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++) {
+            float d = src[n] * M_PI_2;
+
+            dst[n] = sinf(d + contrast * sinf(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,
+};

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);

libavfilter/version.h

@@ -30,7 +30,7 @@
 #include "libavutil/version.h"
 
 #define LIBAVFILTER_VERSION_MAJOR   7
-#define LIBAVFILTER_VERSION_MINOR   1
+#define LIBAVFILTER_VERSION_MINOR   2
 #define LIBAVFILTER_VERSION_MICRO 100
 
 #define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \