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authorPaul B Mahol <onemda@gmail.com>2019-06-30 16:35:07 +0200
committerPaul B Mahol <onemda@gmail.com>2019-07-02 19:02:54 +0200
commitbd5e92ef8ad37d86091ce9d1d89c7cc0f7221ec1 (patch)
treeec59f4fa7033e08cd897cd0bf264738aecefe317
parent1b262004d35a7cec8540fa47a0533dfb12870f95 (diff)
avfilter: add deesser audio filter
-rw-r--r--Changelog1
-rw-r--r--doc/filters.texi36
-rw-r--r--libavfilter/Makefile1
-rw-r--r--libavfilter/af_deesser.c244
-rw-r--r--libavfilter/allfilters.c1
-rw-r--r--libavfilter/version.h2
6 files changed, 284 insertions, 1 deletions
diff --git a/Changelog b/Changelog
index 57476c31b0..86167b76a1 100644
--- a/Changelog
+++ b/Changelog
@@ -34,6 +34,7 @@ version <next>:
- VP4 video decoder
- IFV demuxer
- derain filter
+- deesser filter
version 4.1:
diff --git a/doc/filters.texi b/doc/filters.texi
index 7c355fb1c5..700a76f239 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -3045,6 +3045,42 @@ Optional. It should have a value much less than 1 (e.g. 0.05 or 0.02) and is
used to prevent clipping.
@end table
+@section deesser
+
+Apply de-essing to the audio samples.
+
+@table @option
+@item i
+Set intensity for triggering de-essing. Allowed range is from 0 to 1.
+Default is 0.
+
+@item m
+Set amount of ducking on treble part of sound. Allowed range is from 0 to 1.
+Default is 0.5.
+
+@item f
+How much of original frequency content to keep when de-essing. Allowed range is from 0 to 1.
+Default is 0.5.
+
+@item s
+Set the output mode.
+
+It accepts the following values:
+@table @option
+@item i
+Pass input unchanged.
+
+@item o
+Pass ess filtered out.
+
+@item e
+Pass only ess.
+
+Default value is @var{o}.
+@end table
+
+@end table
+
@section drmeter
Measure audio dynamic range.
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 07ea8d7edc..455c809b15 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -101,6 +101,7 @@ OBJS-$(CONFIG_COMPENSATIONDELAY_FILTER) += af_compensationdelay.o
OBJS-$(CONFIG_CROSSFEED_FILTER) += af_crossfeed.o
OBJS-$(CONFIG_CRYSTALIZER_FILTER) += af_crystalizer.o
OBJS-$(CONFIG_DCSHIFT_FILTER) += af_dcshift.o
+OBJS-$(CONFIG_DEESSER_FILTER) += af_deesser.o
OBJS-$(CONFIG_DRMETER_FILTER) += af_drmeter.o
OBJS-$(CONFIG_DYNAUDNORM_FILTER) += af_dynaudnorm.o
OBJS-$(CONFIG_EARWAX_FILTER) += af_earwax.o
diff --git a/libavfilter/af_deesser.c b/libavfilter/af_deesser.c
new file mode 100644
index 0000000000..2f57d51226
--- /dev/null
+++ b/libavfilter/af_deesser.c
@@ -0,0 +1,244 @@
+/*
+ * Copyright (c) 2018 Chris Johnson
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy
+ * of this software and associated documentation files (the "Software"), to deal
+ * in the Software without restriction, including without limitation the rights
+ * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+ * copies of the Software, and to permit persons to whom the Software is
+ * furnished to do so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in all
+ * copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
+ * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
+ * SOFTWARE.
+ */
+
+#include "libavutil/channel_layout.h"
+#include "libavutil/opt.h"
+#include "avfilter.h"
+#include "audio.h"
+#include "formats.h"
+
+typedef struct DeesserChannel {
+ double s1, s2, s3;
+ double m1, m2;
+ double ratioA, ratioB;
+ double iirSampleA, iirSampleB;
+ int flip;
+} DeesserChannel;
+
+typedef struct DeesserContext {
+ const AVClass *class;
+
+ double intensity;
+ double max;
+ double frequency;
+ int mode;
+
+ DeesserChannel *chan;
+} DeesserContext;
+
+enum OutModes {
+ IN_MODE,
+ OUT_MODE,
+ ESS_MODE,
+ NB_MODES
+};
+
+#define OFFSET(x) offsetof(DeesserContext, x)
+#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
+
+static const AVOption deesser_options[] = {
+ { "i", "set intensity", OFFSET(intensity), AV_OPT_TYPE_DOUBLE, {.dbl=0.0}, 0.0, 1.0, A },
+ { "m", "set max deessing", OFFSET(max), AV_OPT_TYPE_DOUBLE, {.dbl=0.5}, 0.0, 1.0, A },
+ { "f", "set frequency", OFFSET(frequency), AV_OPT_TYPE_DOUBLE, {.dbl=0.5}, 0.0, 1.0, A },
+ { "s", "set output mode", OFFSET(mode), AV_OPT_TYPE_INT, {.i64=OUT_MODE}, 0, NB_MODES-1, A, "mode" },
+ { "i", "input", 0, AV_OPT_TYPE_CONST, {.i64=IN_MODE}, 0, 0, A, "mode" },
+ { "o", "output", 0, AV_OPT_TYPE_CONST, {.i64=OUT_MODE}, 0, 0, A, "mode" },
+ { "e", "ess", 0, AV_OPT_TYPE_CONST, {.i64=ESS_MODE}, 0, 0, A, "mode" },
+ { NULL }
+};
+
+AVFILTER_DEFINE_CLASS(deesser);
+
+static int query_formats(AVFilterContext *ctx)
+{
+ AVFilterFormats *formats = NULL;
+ AVFilterChannelLayouts *layouts = NULL;
+ static const enum AVSampleFormat sample_fmts[] = {
+ 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 int config_input(AVFilterLink *inlink)
+{
+ AVFilterContext *ctx = inlink->dst;
+ DeesserContext *s = ctx->priv;
+
+ s->chan = av_calloc(inlink->channels, sizeof(*s->chan));
+ if (!s->chan)
+ return AVERROR(ENOMEM);
+
+ for (int i = 0; i < inlink->channels; i++) {
+ DeesserChannel *chan = &s->chan[i];;
+
+ chan->ratioA = chan->ratioB = 1.0;
+ }
+
+ return 0;
+}
+
+static int filter_frame(AVFilterLink *inlink, AVFrame *in)
+{
+ AVFilterContext *ctx = inlink->dst;
+ AVFilterLink *outlink = ctx->outputs[0];
+ DeesserContext *s = ctx->priv;
+ AVFrame *out;
+
+ if (av_frame_is_writable(in)) {
+ out = in;
+ } else {
+ out = ff_get_audio_buffer(outlink, in->nb_samples);
+ if (!out) {
+ av_frame_free(&in);
+ return AVERROR(ENOMEM);
+ }
+ av_frame_copy_props(out, in);
+ }
+
+ for (int ch = 0; ch < inlink->channels; ch++) {
+ DeesserChannel *dec = &s->chan[ch];
+ double *src = (double *)in->extended_data[ch];
+ double *dst = (double *)out->extended_data[ch];
+ double overallscale = inlink->sample_rate < 44100 ? 44100.0 / inlink->sample_rate : inlink->sample_rate / 44100.0;
+ double intensity = pow(s->intensity, 5) * (8192 / overallscale);
+ double maxdess = 1.0 / pow(10.0, ((s->max - 1.0) * 48.0) / 20);
+ double iirAmount = pow(s->frequency, 2) / overallscale;
+ double offset;
+ double sense;
+ double recovery;
+ double attackspeed;
+
+ for (int i = 0; i < in->nb_samples; i++) {
+ double sample = src[i];
+
+ dec->s3 = dec->s2;
+ dec->s2 = dec->s1;
+ dec->s1 = sample;
+ dec->m1 = (dec->s1 - dec->s2) * ((dec->s1 - dec->s2) / 1.3);
+ dec->m2 = (dec->s2 - dec->s3) * ((dec->s1 - dec->s2) / 1.3);
+ sense = (dec->m1 - dec->m2) * ((dec->m1 - dec->m2) / 1.3);
+ attackspeed = 7.0 + sense * 1024;
+
+ sense = 1.0 + intensity * intensity * sense;
+ sense = FFMIN(sense, intensity);
+ recovery = 1.0 + (0.01 / sense);
+
+ offset = 1.0 - fabs(sample);
+
+ if (dec->flip) {
+ dec->iirSampleA = (dec->iirSampleA * (1.0 - (offset * iirAmount))) +
+ (sample * (offset * iirAmount));
+ if (dec->ratioA < sense) {
+ dec->ratioA = ((dec->ratioA * attackspeed) + sense) / (attackspeed + 1.0);
+ } else {
+ dec->ratioA = 1.0 + ((dec->ratioA - 1.0) / recovery);
+ }
+
+ dec->ratioA = FFMIN(dec->ratioA, maxdess);
+ sample = dec->iirSampleA + ((sample - dec->iirSampleA) / dec->ratioA);
+ } else {
+ dec->iirSampleB = (dec->iirSampleB * (1.0 - (offset * iirAmount))) +
+ (sample * (offset * iirAmount));
+ if (dec->ratioB < sense) {
+ dec->ratioB = ((dec->ratioB * attackspeed) + sense) / (attackspeed + 1.0);
+ } else {
+ dec->ratioB = 1.0 + ((dec->ratioB - 1.0) / recovery);
+ }
+
+ dec->ratioB = FFMIN(dec->ratioB, maxdess);
+ sample = dec->iirSampleB + ((sample - dec->iirSampleB) / dec->ratioB);
+ }
+
+ dec->flip = !dec->flip;
+
+ if (ctx->is_disabled)
+ sample = src[i];
+
+ switch (s->mode) {
+ case IN_MODE: dst[i] = src[i]; break;
+ case OUT_MODE: dst[i] = sample; break;
+ case ESS_MODE: dst[i] = src[i] - sample; break;
+ }
+ }
+ }
+
+ if (out != in)
+ av_frame_free(&in);
+
+ return ff_filter_frame(outlink, out);
+}
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+ DeesserContext *s = ctx->priv;
+
+ av_freep(&s->chan);
+}
+
+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_deesser = {
+ .name = "deesser",
+ .description = NULL_IF_CONFIG_SMALL("Apply de-essing to the audio."),
+ .query_formats = query_formats,
+ .priv_size = sizeof(DeesserContext),
+ .priv_class = &deesser_class,
+ .uninit = uninit,
+ .inputs = inputs,
+ .outputs = outputs,
+ .flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL,
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index 9c846b1ddd..04a3df7d56 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -93,6 +93,7 @@ extern AVFilter ff_af_compensationdelay;
extern AVFilter ff_af_crossfeed;
extern AVFilter ff_af_crystalizer;
extern AVFilter ff_af_dcshift;
+extern AVFilter ff_af_deesser;
extern AVFilter ff_af_drmeter;
extern AVFilter ff_af_dynaudnorm;
extern AVFilter ff_af_earwax;
diff --git a/libavfilter/version.h b/libavfilter/version.h
index 91a5a1604e..1347e4c7b7 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 55
+#define LIBAVFILTER_VERSION_MINOR 56
#define LIBAVFILTER_VERSION_MICRO 100
#define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \