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-rw-r--r--Changelog1
-rw-r--r--doc/filters.texi10
-rw-r--r--libavfilter/Makefile1
-rw-r--r--libavfilter/af_acontrast.c219
-rw-r--r--libavfilter/allfilters.c1
-rw-r--r--libavfilter/version.h2
6 files changed, 233 insertions, 1 deletions
diff --git a/Changelog b/Changelog
index cda59166fc..e1f8a648b0 100644
--- a/Changelog
+++ b/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:
diff --git a/doc/filters.texi b/doc/filters.texi
index 5d99437871..63ce899784 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 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
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..8b45bd5b2b
--- /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[] = {
+ { "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,
+};
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);
diff --git a/libavfilter/version.h b/libavfilter/version.h
index 33d9ad7bc7..d8484e4263 100644
--- a/libavfilter/version.h
+++ b/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, \
generated by cgit v1.2.3 (git 2.39.1) at 2024-06-28 14:38:26 +0000