[FFmpeg-cvslog] avfilter: add audio dynamic equalizer filter
Paul B Mahol
git at videolan.org
Sun Dec 12 11:51:37 EET 2021
ffmpeg | branch: master | Paul B Mahol <onemda at gmail.com> | Fri Nov 26 14:23:16 2021 +0100| [996b13fac4810efc35ff988f523f0c88a3b57ec9] | committer: Paul B Mahol
avfilter: add audio dynamic equalizer filter
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=996b13fac4810efc35ff988f523f0c88a3b57ec9
---
Changelog | 1 +
doc/filters.texi | 76 +++++++++
libavfilter/Makefile | 1 +
libavfilter/af_adynamicequalizer.c | 315 +++++++++++++++++++++++++++++++++++++
libavfilter/allfilters.c | 1 +
libavfilter/version.h | 2 +-
6 files changed, 395 insertions(+), 1 deletion(-)
diff --git a/Changelog b/Changelog
index 13b22f6149..5c6adc0fa7 100644
--- a/Changelog
+++ b/Changelog
@@ -40,6 +40,7 @@ version <next>:
- adynamicsmooth audio filter
- libplacebo filter
- vflip_vulkan, hflip_vulkan and flip_vulkan filters
+- adynamicequalizer audio filter
version 4.4:
diff --git a/doc/filters.texi b/doc/filters.texi
index 8eff460cd9..cffec8168c 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -843,6 +843,82 @@ Compute derivative/integral of audio stream.
Applying both filters one after another produces original audio.
+ at section adynamicequalizer
+
+Apply dynamic equalization to input audio stream.
+
+A description of the accepted options follows.
+
+ at table @option
+ at item threshold
+Set the detection threshold used to trigger equalization.
+Threshold detection is using bandpass filter.
+Default value is 0. Allowed range is from 0 to 100.
+
+ at item dfrequency
+Set the detection frequency in Hz used for bandpass filter used to trigger equalization.
+Default value is 1000 Hz. Allowed range is between 2 and 1000000 Hz.
+
+ at item dqfactor
+Set the detection resonance factor for bandpass filter used to trigger equalization.
+Default value is 1. Allowed range is from 0.001 to 1000.
+
+ at item tfrequency
+Set the target frequency of equalization filter.
+Default value is 1000 Hz. Allowed range is between 2 and 1000000 Hz.
+
+ at item tqfactor
+Set the target resonance factor for target equalization filter.
+Default value is 1. Allowed range is from 0.001 to 1000.
+
+ at item attack
+Set the amount of milliseconds the signal from detection has to rise above
+the detection threshold before equalization starts.
+Default is 20. Allowed range is between 1 and 2000.
+
+ at item release
+Set the amount of milliseconds the signal from detection has to fall below the
+detection threshold before equalization ends.
+Default is 200. Allowed range is between 1 and 2000.
+
+ at item knee
+Curve the sharp knee around the detection threshold to calculate
+equalization gain more softly.
+Default is 1. Allowed range is between 0 and 8.
+
+ at item ratio
+Set the ratio by which the equalization gain is raised.
+Default is 1. Allowed range is between 1 and 20.
+
+ at item makeup
+Set the makeup offset in dB by which the equalization gain is raised.
+Default is 0. Allowed range is between 0 and 30.
+
+ at item range
+Set the max allowed cut/boost amount in dB. Default is 0.
+Allowed range is from 0 to 200.
+
+ at item slew
+Set the slew factor. Default is 1. Allowed range is from 1 to 200.
+
+ at item mode
+Set the mode of filter operation, can be one of the following:
+
+ at table @samp
+ at item listen
+Output only isolated bandpass signal.
+ at item cut
+Cut frequencies above detection threshold.
+ at item boost
+Boost frequencies bellow detection threshold.
+ at end table
+Default mode is @samp{cut}.
+ at end table
+
+ at subsection Commands
+
+This filter supports the all above options as @ref{commands}.
+
@section adynamicsmooth
Apply dynamic smoothing to input audio stream.
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 8744cc3c63..2fe495df28 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -44,6 +44,7 @@ OBJS-$(CONFIG_ADECORRELATE_FILTER) += af_adecorrelate.o
OBJS-$(CONFIG_ADELAY_FILTER) += af_adelay.o
OBJS-$(CONFIG_ADENORM_FILTER) += af_adenorm.o
OBJS-$(CONFIG_ADERIVATIVE_FILTER) += af_aderivative.o
+OBJS-$(CONFIG_ADYNAMICEQUALIZER_FILTER) += af_adynamicequalizer.o
OBJS-$(CONFIG_ADYNAMICSMOOTH_FILTER) += af_adynamicsmooth.o
OBJS-$(CONFIG_AECHO_FILTER) += af_aecho.o
OBJS-$(CONFIG_AEMPHASIS_FILTER) += af_aemphasis.o
diff --git a/libavfilter/af_adynamicequalizer.c b/libavfilter/af_adynamicequalizer.c
new file mode 100644
index 0000000000..f377a5db3d
--- /dev/null
+++ b/libavfilter/af_adynamicequalizer.c
@@ -0,0 +1,315 @@
+/*
+ * 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 <float.h>
+
+#include "libavutil/opt.h"
+#include "avfilter.h"
+#include "audio.h"
+#include "formats.h"
+#include "hermite.h"
+
+typedef struct AudioDynamicEqualizerContext {
+ const AVClass *class;
+
+ double threshold;
+ double dfrequency;
+ double dqfactor;
+ double tfrequency;
+ double tqfactor;
+ double ratio;
+ double range;
+ double makeup;
+ double knee;
+ double slew;
+ double attack;
+ double release;
+ double attack_coef;
+ double release_coef;
+ int mode;
+
+ AVFrame *state;
+} AudioDynamicEqualizerContext;
+
+static int config_input(AVFilterLink *inlink)
+{
+ AVFilterContext *ctx = inlink->dst;
+ AudioDynamicEqualizerContext *s = ctx->priv;
+
+ s->state = ff_get_audio_buffer(inlink, 8);
+ if (!s->state)
+ return AVERROR(ENOMEM);
+
+ return 0;
+}
+
+static double get_svf(double in, double *m, double *a, double *b)
+{
+ const double v0 = in;
+ const double v3 = v0 - b[1];
+ const double v1 = a[0] * b[0] + a[1] * v3;
+ const double v2 = b[1] + a[1] * b[0] + a[2] * v3;
+
+ b[0] = 2. * v1 - b[0];
+ b[1] = 2. * v2 - b[1];
+
+ return m[0] * v0 + m[1] * v1 + m[2] * v2;
+}
+
+static inline double from_dB(double x)
+{
+ return exp(0.05 * x * M_LN10);
+}
+
+static inline double to_dB(double x)
+{
+ return 20. * log10(x);
+}
+
+static inline double sqr(double x)
+{
+ return x * x;
+}
+
+static double get_gain(double in, double srate, double makeup,
+ double aattack, double iratio, double knee, double range,
+ double thresdb, double slewfactor, double *state,
+ double attack_coeff, double release_coeff, double nc)
+{
+ double width = (6. * knee) + 0.01;
+ double cdb = 0.;
+ double Lgain = 1.;
+ double Lxg, Lxl, Lyg, Lyl, Ly1;
+ double checkwidth = 0.;
+ double slewwidth = 1.8;
+ int attslew = 0;
+
+ Lyg = 0.;
+ Lxg = to_dB(fabs(in) + DBL_EPSILON);
+
+ Lyg = Lxg + (iratio - 1.) * sqr(Lxg - thresdb + width * .5) / (2. * width);
+
+ checkwidth = 2. * fabs(Lxg - thresdb);
+ if (2. * (Lxg - thresdb) < -width) {
+ Lyg = Lxg;
+ } else if (checkwidth <= width) {
+ Lyg = thresdb + (Lxg - thresdb) * iratio;
+ if (checkwidth <= slewwidth) {
+ if (Lyg >= state[2])
+ attslew = 1;
+ }
+ } else if (2. * (Lxg-thresdb) > width) {
+ Lyg = thresdb + (Lxg - thresdb) * iratio;
+ }
+
+ attack_coeff = attslew ? aattack : attack_coeff;
+
+ Lxl = Lxg - Lyg;
+
+ Ly1 = fmaxf(Lxl, release_coeff * state[1] +(1. - release_coeff) * Lxl);
+ Lyl = attack_coeff * state[0] + (1. - attack_coeff) * Ly1;
+
+ cdb = -Lyl;
+ Lgain = from_dB(nc * fmin(cdb - makeup, range));
+
+ state[0] = Lyl;
+ state[1] = Ly1;
+ state[2] = Lyg;
+
+ return Lgain;
+}
+
+typedef struct ThreadData {
+ AVFrame *in, *out;
+} ThreadData;
+
+static int filter_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
+{
+ AudioDynamicEqualizerContext *s = ctx->priv;
+ ThreadData *td = arg;
+ AVFrame *in = td->in;
+ AVFrame *out = td->out;
+ const double sample_rate = in->sample_rate;
+ const double makeup = s->makeup;
+ const double iratio = 1. / s->ratio;
+ const double range = s->range;
+ const double dfrequency = fmin(s->dfrequency, sample_rate * 0.5);
+ const double tfrequency = fmin(s->tfrequency, sample_rate * 0.5);
+ const double threshold = log(s->threshold + DBL_EPSILON);
+ const double release = s->release_coef;
+ const double attack = s->attack_coef;
+ const double dqfactor = s->dqfactor;
+ const double tqfactor = s->tqfactor;
+ const double fg = tan(M_PI * tfrequency / sample_rate);
+ const double dg = tan(M_PI * dfrequency / sample_rate);
+ const int start = (in->channels * jobnr) / nb_jobs;
+ const int end = (in->channels * (jobnr+1)) / nb_jobs;
+ const int mode = s->mode;
+ const double knee = s->knee;
+ const double slew = s->slew;
+ const double aattack = exp(-1000. / ((s->attack + 2.0 * (slew - 1.)) * sample_rate));
+ const double nc = mode == 0 ? 1. : -1.;
+ double da[3], dm[3];
+
+ {
+ double k = 1. / dqfactor;
+
+ da[0] = 1. / (1. + dg * (dg + k));
+ da[1] = dg * da[0];
+ da[2] = dg * da[1];
+
+ dm[0] = 0.;
+ dm[1] = 1.;
+ dm[2] = 0.;
+ }
+
+ for (int ch = start; ch < end; ch++) {
+ const double *src = (const double *)in->extended_data[ch];
+ double *dst = (double *)out->extended_data[ch];
+ double *state = (double *)s->state->extended_data[ch];
+
+ for (int n = 0; n < out->nb_samples; n++) {
+ double detect, gain, v, listen;
+ double fa[3], fm[3];
+
+ detect = listen = get_svf(src[n], dm, da, state);
+ detect = fabs(detect);
+
+ gain = get_gain(detect, sample_rate, makeup,
+ aattack, iratio, knee, range, threshold, slew,
+ &state[4], attack, release, nc);
+
+ {
+ double k = 1. / (tqfactor * gain);
+
+ fa[0] = 1. / (1. + fg * (fg + k));
+ fa[1] = fg * fa[0];
+ fa[2] = fg * fa[1];
+
+ fm[0] = 1.;
+ fm[1] = k * (gain * gain - 1.);
+ fm[2] = 0.;
+ }
+
+ v = get_svf(src[n], fm, fa, &state[2]);
+ v = mode == -1 ? listen : v;
+ dst[n] = ctx->is_disabled ? src[n] : v;
+ }
+ }
+
+ return 0;
+}
+
+static double get_coef(double x, double sr)
+{
+ return exp(-1000. / (x * sr));
+}
+
+static int filter_frame(AVFilterLink *inlink, AVFrame *in)
+{
+ AVFilterContext *ctx = inlink->dst;
+ AVFilterLink *outlink = ctx->outputs[0];
+ AudioDynamicEqualizerContext *s = ctx->priv;
+ ThreadData td;
+ 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);
+ }
+
+ s->attack_coef = get_coef(s->attack, in->sample_rate);
+ s->release_coef = get_coef(s->release, in->sample_rate);
+
+ td.in = in;
+ td.out = out;
+ ff_filter_execute(ctx, filter_channels, &td, NULL,
+ FFMIN(outlink->channels, ff_filter_get_nb_threads(ctx)));
+
+ if (out != in)
+ av_frame_free(&in);
+ return ff_filter_frame(outlink, out);
+}
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+ AudioDynamicEqualizerContext *s = ctx->priv;
+
+ av_frame_free(&s->state);
+}
+
+#define OFFSET(x) offsetof(AudioDynamicEqualizerContext, x)
+#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
+
+static const AVOption adynamicequalizer_options[] = {
+ { "threshold", "set detection threshold", OFFSET(threshold), AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0, 100, FLAGS },
+ { "dfrequency", "set detection frequency", OFFSET(dfrequency), AV_OPT_TYPE_DOUBLE, {.dbl=1000}, 2, 1000000, FLAGS },
+ { "dqfactor", "set detection Q factor", OFFSET(dqfactor), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.001, 1000, FLAGS },
+ { "tfrequency", "set target frequency", OFFSET(tfrequency), AV_OPT_TYPE_DOUBLE, {.dbl=1000}, 2, 1000000, FLAGS },
+ { "tqfactor", "set target Q factor", OFFSET(tqfactor), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.001, 1000, FLAGS },
+ { "attack", "set attack duration", OFFSET(attack), AV_OPT_TYPE_DOUBLE, {.dbl=20}, 1, 2000, FLAGS },
+ { "release", "set release duration", OFFSET(release), AV_OPT_TYPE_DOUBLE, {.dbl=200}, 1, 2000, FLAGS },
+ { "knee", "set knee factor", OFFSET(knee), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 8, FLAGS },
+ { "ratio", "set ratio factor", OFFSET(ratio), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 1, 20, FLAGS },
+ { "makeup", "set makeup gain", OFFSET(makeup), AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0, 30, FLAGS },
+ { "range", "set max gain", OFFSET(range), AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0, 200, FLAGS },
+ { "slew", "set slew factor", OFFSET(slew), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 1, 200, FLAGS },
+ { "mode", "set mode", OFFSET(mode), AV_OPT_TYPE_INT, {.i64=0}, -1, 1, FLAGS, "mode" },
+ { "listen", 0, 0, AV_OPT_TYPE_CONST, {.i64=-1}, 0, 0, FLAGS, "mode" },
+ { "cut", 0, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, FLAGS, "mode" },
+ { "boost", 0, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, FLAGS, "mode" },
+ { NULL }
+};
+
+AVFILTER_DEFINE_CLASS(adynamicequalizer);
+
+static const AVFilterPad inputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .filter_frame = filter_frame,
+ .config_props = config_input,
+ },
+};
+
+static const AVFilterPad outputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ },
+};
+
+const AVFilter ff_af_adynamicequalizer = {
+ .name = "adynamicequalizer",
+ .description = NULL_IF_CONFIG_SMALL("Apply Dynamic Equalization of input audio."),
+ .priv_size = sizeof(AudioDynamicEqualizerContext),
+ .priv_class = &adynamicequalizer_class,
+ .uninit = uninit,
+ FILTER_INPUTS(inputs),
+ FILTER_OUTPUTS(outputs),
+ FILTER_SINGLE_SAMPLEFMT(AV_SAMPLE_FMT_DBLP),
+ .flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL |
+ AVFILTER_FLAG_SLICE_THREADS,
+ .process_command = ff_filter_process_command,
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index 9e16b4e71e..ec57a2c49c 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -37,6 +37,7 @@ extern const AVFilter ff_af_adecorrelate;
extern const AVFilter ff_af_adelay;
extern const AVFilter ff_af_adenorm;
extern const AVFilter ff_af_aderivative;
+extern const AVFilter ff_af_adynamicequalizer;
extern const AVFilter ff_af_adynamicsmooth;
extern const AVFilter ff_af_aecho;
extern const AVFilter ff_af_aemphasis;
diff --git a/libavfilter/version.h b/libavfilter/version.h
index 0247fb4f9a..0253c911be 100644
--- a/libavfilter/version.h
+++ b/libavfilter/version.h
@@ -30,7 +30,7 @@
#include "libavutil/version.h"
#define LIBAVFILTER_VERSION_MAJOR 8
-#define LIBAVFILTER_VERSION_MINOR 19
+#define LIBAVFILTER_VERSION_MINOR 20
#define LIBAVFILTER_VERSION_MICRO 100
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