[FFmpeg-devel] [PATCH] avfilter: add acomb filter
Paul B Mahol
onemda at gmail.com
Wed Oct 2 18:11:29 EEST 2019
Signed-off-by: Paul B Mahol <onemda at gmail.com>
---
doc/filters.texi | 28 ++++++
libavfilter/Makefile | 1 +
libavfilter/af_acomb.c | 188 +++++++++++++++++++++++++++++++++++++++
libavfilter/allfilters.c | 1 +
4 files changed, 218 insertions(+)
create mode 100644 libavfilter/af_acomb.c
diff --git a/doc/filters.texi b/doc/filters.texi
index e46839bfec..9c50b2e4b2 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -355,6 +355,34 @@ build.
Below is a description of the currently available audio filters.
+ at section acomb
+Apply comb audio filtering.
+
+Amplifies or attenuates certain frequencies by the superposition of a
+delayed version of the original audio signal onto itself.
+
+ at table @option
+ at item t
+Set comb filtering type.
+
+It accepts the following values:
+ at table @option
+ at item f
+set feedforward type
+ at item b
+set feedback type
+ at end table
+
+ at item b0
+Set direct signal gain. Default is 1. Allowed range is from 0 to 1.
+
+ at item xM
+Set delayed line gain. Default is 0.5. Allowed range is from 0 to 1.
+
+ at item M
+Set delay in number of samples. Default is 10. Allowed range is from 1 to 327680.
+ at end table
+
@section acompressor
A compressor is mainly used to reduce the dynamic range of a signal.
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 182fe9df4b..d8a16d6e15 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -31,6 +31,7 @@ include $(SRC_PATH)/libavfilter/dnn/Makefile
# audio filters
OBJS-$(CONFIG_ABENCH_FILTER) += f_bench.o
+OBJS-$(CONFIG_ACOMB_FILTER) += af_acomb.o
OBJS-$(CONFIG_ACOMPRESSOR_FILTER) += af_sidechaincompress.o
OBJS-$(CONFIG_ACONTRAST_FILTER) += af_acontrast.o
OBJS-$(CONFIG_ACOPY_FILTER) += af_acopy.o
diff --git a/libavfilter/af_acomb.c b/libavfilter/af_acomb.c
new file mode 100644
index 0000000000..3b0730c363
--- /dev/null
+++ b/libavfilter/af_acomb.c
@@ -0,0 +1,188 @@
+/*
+ * 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/opt.h"
+#include "audio.h"
+#include "avfilter.h"
+#include "internal.h"
+
+typedef struct AudioCombContext {
+ const AVClass *class;
+
+ double b0, xM;
+ int t, M;
+
+ int head;
+ int tail;
+
+ AVFrame *delayframe;
+
+ void (*filter)(struct AudioCombContext *s, AVFrame *in, AVFrame *out);
+} AudioCombContext;
+
+#define OFFSET(x) offsetof(AudioCombContext, x)
+#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
+
+static const AVOption acomb_options[] = {
+ { "t", "set comb filter type", OFFSET(t), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, A, "t" },
+ { "f", "feedforward", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, A, "t" },
+ { "b", "feedback", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, A, "t" },
+ { "b0", "set direct signal gain", OFFSET(b0), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 1, A },
+ { "xM", "set delayed line gain", OFFSET(xM), AV_OPT_TYPE_DOUBLE, {.dbl=0.5}, 0, 1, A },
+ { "M", "set delay in number of samples", OFFSET(M), AV_OPT_TYPE_INT, {.i64=10}, 1, 327680, A },
+ { NULL }
+};
+
+AVFILTER_DEFINE_CLASS(acomb);
+
+static int query_formats(AVFilterContext *ctx)
+{
+ AVFilterFormats *formats = NULL;
+ AVFilterChannelLayouts *layouts = NULL;
+ static const enum AVSampleFormat sample_fmts[] = {
+ AV_SAMPLE_FMT_FLTP,
+ 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);
+}
+
+#define COMB(name, type, dir, t) \
+static void acomb_## name ## _ ##dir(AudioCombContext *s, \
+ AVFrame *in, AVFrame *out) \
+{ \
+ const type b0 = s->b0; \
+ const type xM = s->xM; \
+ const int M = s->M; \
+ int head; \
+ \
+ for (int c = 0; c < in->channels; c++) { \
+ const type *src = (const type *)in->extended_data[c]; \
+ type *delay = (type *)s->delayframe->extended_data[c]; \
+ type *dst = (type *)out->extended_data[c]; \
+ \
+ head = s->head; \
+ for (int n = 0; n < in->nb_samples; n++) { \
+ dst[n] = b0 * src[n] + t * xM * delay[head]; \
+ if (t == 1) \
+ delay[head] = src[n]; \
+ else \
+ delay[head] = dst[n]; \
+ head++; \
+ if (head >= M) \
+ head = 0; \
+ } \
+ } \
+ \
+ s->head = head; \
+}
+
+COMB(fltp, float, f, 1)
+COMB(dblp, double, f, 1)
+COMB(fltp, float, b, -1)
+COMB(dblp, double, b, -1)
+
+static int config_input(AVFilterLink *inlink)
+{
+ AVFilterContext *ctx = inlink->dst;
+ AudioCombContext *s = ctx->priv;
+
+ s->delayframe = ff_get_audio_buffer(inlink, s->M);
+ if (!s->delayframe)
+ return AVERROR(ENOMEM);
+
+ switch (inlink->format) {
+ case AV_SAMPLE_FMT_FLTP: s->filter = s->t ? acomb_fltp_b : acomb_fltp_f; break;
+ case AV_SAMPLE_FMT_DBLP: s->filter = s->t ? acomb_dblp_b : acomb_dblp_f; break;
+ }
+
+ return 0;
+}
+
+static int filter_frame(AVFilterLink *inlink, AVFrame *in)
+{
+ AVFilterContext *ctx = inlink->dst;
+ AudioCombContext *s = ctx->priv;
+ AVFilterLink *outlink = ctx->outputs[0];
+ AVFrame *out = ff_get_audio_buffer(outlink, in->nb_samples);
+
+ if (!out) {
+ av_frame_free(&in);
+ return AVERROR(ENOMEM);
+ }
+ av_frame_copy_props(out, in);
+
+ s->filter(s, in, out);
+
+ av_frame_free(&in);
+ return ff_filter_frame(outlink, out);
+}
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+ AudioCombContext *s = ctx->priv;
+
+ av_frame_free(&s->delayframe);
+}
+
+static const AVFilterPad acomb_inputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .filter_frame = filter_frame,
+ .config_props = config_input,
+ },
+ { NULL }
+};
+
+static const AVFilterPad acomb_outputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ },
+ { NULL }
+};
+
+AVFilter ff_af_acomb = {
+ .name = "acomb",
+ .description = NULL_IF_CONFIG_SMALL("Apply comb audio filter."),
+ .query_formats = query_formats,
+ .priv_size = sizeof(AudioCombContext),
+ .priv_class = &acomb_class,
+ .uninit = uninit,
+ .inputs = acomb_inputs,
+ .outputs = acomb_outputs,
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index 1a26129069..7417f9656d 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -24,6 +24,7 @@
#include "config.h"
extern AVFilter ff_af_abench;
+extern AVFilter ff_af_acomb;
extern AVFilter ff_af_acompressor;
extern AVFilter ff_af_acontrast;
extern AVFilter ff_af_acopy;
--
2.17.1
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