lavfi: add compand filter

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

6 files changed, 597 insertions, 2 deletions

diff --git a/Changelog b/Changelog
index a8ff9b1..cbee5fa 100644
--- a/Changelog
+++ b/Changelog
@@ -6,6 +6,7 @@ version <next>
 - aecho filter
 - perspective filter ported from libmpcodecs
 - ffprobe -show_programs option
+- compand filter
 
 
 version 2.0:
diff --git a/doc/filters.texi b/doc/filters.texi
index 0c18446..3404716 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -1176,6 +1176,83 @@ front_center.wav -map '[LFE]' lfe.wav -map '[SL]' side_left.wav -map '[SR]'
 side_right.wav
 @end example
 
+@section compand
+
+Compress or expand audio dynamic range.
+
+A description of the accepted options follows.
+
+@table @option
+@item attacks
+@item decays
+Set list of times in seconds for each channel over which the instantaneous
+level of the input signal is averaged to determine its volume.
+@option{attacks} refers to increase of volume and @option{decays} refers
+to decrease of volume.
+For most situations, the attack time (response to the audio getting louder)
+should be shorter than the decay time because the human ear is more sensitive
+to sudden loud audio than sudden soft audio.
+Typical value for attack is @code{0.3} seconds and for decay @code{0.8}
+seconds.
+
+@item points
+Set list of points for transfer function, specified in dB relative to maximum
+possible signal amplitude.
+Each key points list need to be defined using the following syntax:
+@code{x0/y0 x1/y1 x2/y2 ...}.
+
+The input values must be in strictly increasing order but the transfer
+function does not have to me monotonically rising.
+The point @code{0/0} is assumed but may be overridden (by @code{0/out-dBn}).
+Typical values for the transfer function are @code{-70/-70 -60/-20}.
+
+@item soft-knee
+Set amount for which the points at where adjacent line segments on the
+transfer function meet will be rounded. Defaults is @code{0.01}.
+
+@item gain
+Set additional gain in dB to be applied at all points on the transfer function
+and allows easy adjustment of the overall gain.
+Default is @code{0}.
+
+@item volume
+Set initial volume in dB to be assumed for each channel when filtering starts.
+This permits the user to supply a nominal level initially, so that,
+for example, a very large gain is not applied to initial signal levels before
+the companding has begun to operate. A typical value for audio which is
+initially quiet is -90 dB. Default is @code{0}.
+
+@item delay
+Set delay in seconds. Default is @code{0}. The input audio
+is analysed immediately, but audio is delayed before being fed to the
+volume adjuster. Specifying a delay approximately equal to the attack/decay
+times allows the filter to effectively operate in predictive rather than
+reactive mode.
+@end table
+
+@subsection Examples
+@itemize
+@item
+Make music with both quiet and loud passages suitable for listening
+in a noisy environment:
+@example
+compand=.3 .3:1 1:-90/-60 -60/-40 -40/-30 -20/-20:6:0:-90:0.2
+@end example
+
+@item
+Noise-gate for when the noise is at a lower level than the signal:
+@example
+compand=.1 .1:.2 .2:-900/-900 -50.1/-900 -50/-50:.01:0:-90:.1
+@end example
+
+@item
+Here is another noise-gate, this time for when the noise is at a higher level
+than the signal (making it, in some ways, similar to squelch):
+@example
+compand=.1 .1:.1 .1:-45.1/-45.1 -45/-900 0/-900:.01:45:-90:.1
+@end example
+@end itemize
+
 @section earwax
 
 Make audio easier to listen to on headphones.
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index f54e100..3751d54 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -84,6 +84,7 @@ OBJS-$(CONFIG_BASS_FILTER)                   += af_biquads.o
 OBJS-$(CONFIG_BIQUAD_FILTER)                 += af_biquads.o
 OBJS-$(CONFIG_CHANNELMAP_FILTER)             += af_channelmap.o
 OBJS-$(CONFIG_CHANNELSPLIT_FILTER)           += af_channelsplit.o
+OBJS-$(CONFIG_COMPAND_FILTER)                += af_compand.o
 OBJS-$(CONFIG_EARWAX_FILTER)                 += af_earwax.o
 OBJS-$(CONFIG_EBUR128_FILTER)                += f_ebur128.o
 OBJS-$(CONFIG_EQUALIZER_FILTER)              += af_biquads.o
diff --git a/libavfilter/af_compand.c b/libavfilter/af_compand.c
new file mode 100644
index 0000000..8630cda
--- a/dev/null
+++ b/libavfilter/af_compand.c
@@ -0,0 +1,515 @@
+/*
+ * Copyright (c) 1999 Chris Bagwell
+ * Copyright (c) 1999 Nick Bailey
+ * Copyright (c) 2007 Rob Sykes <robs@users.sourceforge.net>
+ * Copyright (c) 2013 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/avstring.h"
+#include "libavutil/opt.h"
+#include "libavutil/samplefmt.h"
+#include "avfilter.h"
+#include "audio.h"
+#include "internal.h"
+
+typedef struct ChanParam {
+    double attack;
+    double decay;
+    double volume;
+} ChanParam;
+
+typedef struct CompandSegment {
+    double x, y;
+    double a, b;
+} CompandSegment;
+
+typedef struct CompandContext {
+    const AVClass *class;
+    char *attacks, *decays, *points;
+    CompandSegment *segments;
+    ChanParam *channels;
+    double in_min_lin;
+    double out_min_lin;
+    double curve_dB;
+    double gain_dB;
+    double initial_volume;
+    double delay;
+    uint8_t **delayptrs;
+    int delay_samples;
+    int delay_count;
+    int delay_index;
+    int64_t pts;
+
+    int (*compand)(AVFilterContext *ctx, AVFrame *frame);
+} CompandContext;
+
+#define OFFSET(x) offsetof(CompandContext, x)
+#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
+
+static const AVOption compand_options[] = {
+    { "attacks", "set time over which increase of volume is determined", OFFSET(attacks), AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, A },
+    { "decays", "set time over which decrease of volume is determined", OFFSET(decays), AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, A },
+    { "points", "set points of transfer function", OFFSET(points), AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, A },
+    { "soft-knee", "set soft-knee", OFFSET(curve_dB), AV_OPT_TYPE_DOUBLE, {.dbl=0.01}, 0.01, 900, A },
+    { "gain", "set output gain", OFFSET(gain_dB), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -900, 900, A },
+    { "volume", "set initial volume", OFFSET(initial_volume), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -900, 0, A },
+    { "delay", "set delay for samples before sending them to volume adjuster", OFFSET(delay), AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0, 20, A },
+    { NULL },
+};
+
+AVFILTER_DEFINE_CLASS(compand);
+
+static av_cold int init(AVFilterContext *ctx)
+{
+    CompandContext *s = ctx->priv;
+
+    if (!s->attacks || !s->decays || !s->points) {
+        av_log(ctx, AV_LOG_ERROR, "Missing attacks and/or decays and/or points.\n");
+        return AVERROR(EINVAL);
+    }
+
+    return 0;
+}
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+    CompandContext *s = ctx->priv;
+
+    av_freep(&s->channels);
+    av_freep(&s->segments);
+    if (s->delayptrs)
+        av_freep(&s->delayptrs[0]);
+    av_freep(&s->delayptrs);
+}
+
+static int query_formats(AVFilterContext *ctx)
+{
+    AVFilterChannelLayouts *layouts;
+    AVFilterFormats *formats;
+    static const enum AVSampleFormat sample_fmts[] = {
+        AV_SAMPLE_FMT_DBLP,
+        AV_SAMPLE_FMT_NONE
+    };
+
+    layouts = ff_all_channel_layouts();
+    if (!layouts)
+        return AVERROR(ENOMEM);
+    ff_set_common_channel_layouts(ctx, layouts);
+
+    formats = ff_make_format_list(sample_fmts);
+    if (!formats)
+        return AVERROR(ENOMEM);
+    ff_set_common_formats(ctx, formats);
+
+    formats = ff_all_samplerates();
+    if (!formats)
+        return AVERROR(ENOMEM);
+    ff_set_common_samplerates(ctx, formats);
+
+    return 0;
+}
+
+static void count_items(char *item_str, int *nb_items)
+{
+    char *p;
+
+    *nb_items = 1;
+    for (p = item_str; *p; p++) {
+        if (*p == ' ')
+            (*nb_items)++;
+    }
+
+}
+
+static void update_volume(ChanParam *cp, double in)
+{
+    double delta = in - cp->volume;
+
+    if (delta > 0.0)
+        cp->volume += delta * cp->attack;
+    else
+        cp->volume += delta * cp->decay;
+}
+
+static double get_volume(CompandContext *s, double in_lin)
+{
+    CompandSegment *cs;
+    double in_log, out_log;
+    int i;
+
+    if (in_lin < s->in_min_lin)
+        return s->out_min_lin;
+
+    in_log = log(in_lin);
+
+    for (i = 1;; i++)
+        if (in_log <= s->segments[i + 1].x)
+            break;
+
+    cs = &s->segments[i];
+    in_log -= cs->x;
+    out_log = cs->y + in_log * (cs->a * in_log + cs->b);
+
+    return exp(out_log);
+}
+
+static int compand_nodelay(AVFilterContext *ctx, AVFrame *frame)
+{
+    CompandContext *s = ctx->priv;
+    AVFilterLink *inlink = ctx->inputs[0];
+    const int channels = inlink->channels;
+    const int nb_samples = frame->nb_samples;
+    AVFrame *out_frame;
+    int chan, i;
+
+    if (av_frame_is_writable(frame)) {
+        out_frame = frame;
+    } else {
+        out_frame = ff_get_audio_buffer(inlink, nb_samples);
+        if (!out_frame)
+            return AVERROR(ENOMEM);
+        av_frame_copy_props(out_frame, frame);
+    }
+
+    for (chan = 0; chan < channels; chan++) {
+        const double *src = (double *)frame->data[chan];
+        double *dst = (double *)out_frame->data[chan];
+        ChanParam *cp = &s->channels[chan];
+
+        for (i = 0; i < nb_samples; i++) {
+            update_volume(cp, fabs(src[i]));
+
+            dst[i] = av_clipd(src[i] * get_volume(s, cp->volume), -1, 1);
+        }
+    }
+
+    if (frame != out_frame)
+        av_frame_free(&frame);
+
+    return ff_filter_frame(ctx->outputs[0], out_frame);
+}
+
+#define MOD(a, b) (((a) >= (b)) ? (a) - (b) : (a))
+
+static int compand_delay(AVFilterContext *ctx, AVFrame *frame)
+{
+    CompandContext *s = ctx->priv;
+    AVFilterLink *inlink = ctx->inputs[0];
+    const int channels = inlink->channels;
+    const int nb_samples = frame->nb_samples;
+    int chan, i, dindex, oindex, count;
+    AVFrame *out_frame = NULL;
+
+    for (chan = 0; chan < channels; chan++) {
+        const double *src = (double *)frame->data[chan];
+        double *dbuf = (double *)s->delayptrs[chan];
+        ChanParam *cp = &s->channels[chan];
+        double *dst;
+
+        count  = s->delay_count;
+        dindex = s->delay_index;
+        for (i = 0, oindex = 0; i < nb_samples; i++) {
+            const double in = src[i];
+            update_volume(cp, fabs(in));
+
+            if (count >= s->delay_samples) {
+                if (!out_frame) {
+                    out_frame = ff_get_audio_buffer(inlink, nb_samples - i);
+                    if (!out_frame)
+                        return AVERROR(ENOMEM);
+                    av_frame_copy_props(out_frame, frame);
+                    out_frame->pts = s->pts;
+                    s->pts += av_rescale_q(nb_samples - i, (AVRational){1, inlink->sample_rate}, inlink->time_base);
+                }
+
+                dst = (double *)out_frame->data[chan];
+                dst[oindex++] = av_clipd(dbuf[dindex] * get_volume(s, cp->volume), -1, 1);
+            } else {
+                count++;
+            }
+
+            dbuf[dindex] = in;
+            dindex = MOD(dindex + 1, s->delay_samples);
+        }
+    }
+
+    s->delay_count = count;
+    s->delay_index = dindex;
+
+    av_frame_free(&frame);
+    return out_frame ? ff_filter_frame(ctx->outputs[0], out_frame) : 0;
+}
+
+static int compand_drain(AVFilterLink *outlink)
+{
+    AVFilterContext *ctx = outlink->src;
+    CompandContext *s = ctx->priv;
+    const int channels = outlink->channels;
+    int chan, i, dindex;
+    AVFrame *frame = NULL;
+
+    frame = ff_get_audio_buffer(outlink, FFMIN(2048, s->delay_count));
+    if (!frame)
+        return AVERROR(ENOMEM);
+    frame->pts = s->pts;
+    s->pts += av_rescale_q(frame->nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base);
+
+    for (chan = 0; chan < channels; chan++) {
+        double *dbuf = (double *)s->delayptrs[chan];
+        double *dst = (double *)frame->data[chan];
+        ChanParam *cp = &s->channels[chan];
+
+        dindex = s->delay_index;
+        for (i = 0; i < frame->nb_samples; i++) {
+            dst[i] = av_clipd(dbuf[dindex] * get_volume(s, cp->volume), -1, 1);
+            dindex = MOD(dindex + 1, s->delay_samples);
+        }
+    }
+    s->delay_count -= frame->nb_samples;
+    s->delay_index = dindex;
+
+    return ff_filter_frame(outlink, frame);
+}
+
+static int config_output(AVFilterLink *outlink)
+{
+    AVFilterContext *ctx = outlink->src;
+    CompandContext *s = ctx->priv;
+    const int sample_rate = outlink->sample_rate;
+    double radius = s->curve_dB * M_LN10 / 20;
+    int nb_attacks, nb_decays, nb_points;
+    char *p, *saveptr = NULL;
+    int new_nb_items, num;
+    int i;
+
+    count_items(s->attacks, &nb_attacks);
+    count_items(s->decays, &nb_decays);
+    count_items(s->points, &nb_points);
+
+    if ((nb_attacks > outlink->channels) || (nb_decays > outlink->channels)) {
+        av_log(ctx, AV_LOG_ERROR, "Number of attacks/decays bigger than number of channels.\n");
+        return AVERROR(EINVAL);
+    }
+
+    uninit(ctx);
+
+    s->channels = av_mallocz_array(outlink->channels, sizeof(*s->channels));
+    s->segments = av_mallocz_array((nb_points + 4) * 2, sizeof(*s->segments));
+
+    if (!s->channels || !s->segments)
+        return AVERROR(ENOMEM);
+
+    p = s->attacks;
+    for (i = 0, new_nb_items = 0; i < nb_attacks; i++) {
+        char *tstr = av_strtok(p, " ", &saveptr);
+        p = NULL;
+        new_nb_items += sscanf(tstr, "%lf", &s->channels[i].attack) == 1;
+        if (s->channels[i].attack < 0)
+            return AVERROR(EINVAL);
+    }
+    nb_attacks = new_nb_items;
+
+    p = s->decays;
+    for (i = 0, new_nb_items = 0; i < nb_decays; i++) {
+        char *tstr = av_strtok(p, " ", &saveptr);
+        p = NULL;
+        new_nb_items += sscanf(tstr, "%lf", &s->channels[i].decay) == 1;
+        if (s->channels[i].decay < 0)
+            return AVERROR(EINVAL);
+    }
+    nb_decays = new_nb_items;
+
+    if (nb_attacks != nb_decays) {
+        av_log(ctx, AV_LOG_ERROR, "Number of attacks %d differs from number of decays %d.\n", nb_attacks, nb_decays);
+        return AVERROR(EINVAL);
+    }
+
+#define S(x) s->segments[2 * ((x) + 1)]
+    p = s->points;
+    for (i = 0, new_nb_items = 0; i < nb_points; i++) {
+        char *tstr = av_strtok(p, " ", &saveptr);
+        p = NULL;
+        if (sscanf(tstr, "%lf/%lf", &S(i).x, &S(i).y) != 2) {
+            av_log(ctx, AV_LOG_ERROR, "Invalid and/or missing input/output value.\n");
+            return AVERROR(EINVAL);
+        }
+        if (i && S(i - 1).x > S(i).x) {
+            av_log(ctx, AV_LOG_ERROR, "Transfer function input values must be increasing.\n");
+            return AVERROR(EINVAL);
+        }
+        S(i).y -= S(i).x;
+        av_log(ctx, AV_LOG_DEBUG, "%d: x=%lf y=%lf\n", i, S(i).x, S(i).y);
+        new_nb_items++;
+    }
+    num = new_nb_items;
+
+    /* Add 0,0 if necessary */
+    if (num == 0 || S(num - 1).x)
+        num++;
+
+#undef S
+#define S(x) s->segments[2 * (x)]
+    /* Add a tail off segment at the start */
+    S(0).x = S(1).x - 2 * s->curve_dB;
+    S(0).y = S(1).y;
+    num++;
+
+    /* Join adjacent colinear segments */
+    for (i = 2; i < num; i++) {
+        double g1 = (S(i - 1).y - S(i - 2).y) * (S(i - 0).x - S(i - 1).x);
+        double g2 = (S(i - 0).y - S(i - 1).y) * (S(i - 1).x - S(i - 2).x);
+        int j;
+
+        if (fabs(g1 - g2))
+            continue;
+        num--;
+        for (j = --i; j < num; j++)
+            S(j) = S(j + 1);
+    }
+
+    for (i = 0; !i || s->segments[i - 2].x; i += 2) {
+        s->segments[i].y += s->gain_dB;
+        s->segments[i].x *= M_LN10 / 20;
+        s->segments[i].y *= M_LN10 / 20;
+    }
+
+#define L(x) s->segments[i - (x)]
+    for (i = 4; s->segments[i - 2].x; i += 2) {
+        double x, y, cx, cy, in1, in2, out1, out2, theta, len, r;
+
+        L(4).a = 0;
+        L(4).b = (L(2).y - L(4).y) / (L(2).x - L(4).x);
+
+        L(2).a = 0;
+        L(2).b = (L(0).y - L(2).y) / (L(0).x - L(2).x);
+
+        theta = atan2(L(2).y - L(4).y, L(2).x - L(4).x);
+        len = sqrt(pow(L(2).x - L(4).x, 2.) + pow(L(2).y - L(4).y, 2.));
+        r = FFMIN(radius, len);
+        L(3).x = L(2).x - r * cos(theta);
+        L(3).y = L(2).y - r * sin(theta);
+
+        theta = atan2(L(0).y - L(2).y, L(0).x - L(2).x);
+        len = sqrt(pow(L(0).x - L(2).x, 2.) + pow(L(0).y - L(2).y, 2.));
+        r = FFMIN(radius, len / 2);
+        x = L(2).x + r * cos(theta);
+        y = L(2).y + r * sin(theta);
+
+        cx = (L(3).x + L(2).x + x) / 3;
+        cy = (L(3).y + L(2).y + y) / 3;
+
+        L(2).x = x;
+        L(2).y = y;
+
+        in1 = cx - L(3).x;
+        out1 = cy - L(3).y;
+        in2 = L(2).x - L(3).x;
+        out2 = L(2).y - L(3).y;
+        L(3).a = (out2 / in2 - out1 / in1) / (in2-in1);
+        L(3).b = out1 / in1 - L(3).a * in1;
+    }
+    L(3).x = 0;
+    L(3).y = L(2).y;
+
+    s->in_min_lin  = exp(s->segments[1].x);
+    s->out_min_lin = exp(s->segments[1].y);
+
+    for (i = 0; i < outlink->channels; i++) {
+        ChanParam *cp = &s->channels[i];
+
+        if (cp->attack > 1.0 / sample_rate)
+            cp->attack = 1.0 - exp(-1.0 / (sample_rate * cp->attack));
+        else
+            cp->attack = 1.0;
+        if (cp->decay > 1.0 / sample_rate)
+            cp->decay = 1.0 - exp(-1.0 / (sample_rate * cp->decay));
+        else
+            cp->decay = 1.0;
+        cp->volume = pow(10.0, s->initial_volume / 20);
+    }
+
+    s->delay_samples = s->delay * sample_rate;
+    if (s->delay_samples > 0) {
+        int ret;
+        if ((ret = av_samples_alloc_array_and_samples(&s->delayptrs, NULL,
+                                                      outlink->channels,
+                                                      s->delay_samples,
+                                                      outlink->format, 0)) < 0)
+            return ret;
+        s->compand = compand_delay;
+        outlink->flags |= FF_LINK_FLAG_REQUEST_LOOP;
+    } else {
+        s->compand = compand_nodelay;
+    }
+    return 0;
+}
+
+static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
+{
+    AVFilterContext *ctx = inlink->dst;
+    CompandContext *s = ctx->priv;
+
+    return s->compand(ctx, frame);
+}
+
+static int request_frame(AVFilterLink *outlink)
+{
+    AVFilterContext *ctx = outlink->src;
+    CompandContext *s = ctx->priv;
+    int ret;
+
+    ret = ff_request_frame(ctx->inputs[0]);
+
+    if (ret == AVERROR_EOF && !ctx->is_disabled && s->delay_count)
+        ret = compand_drain(outlink);
+
+    return ret;
+}
+
+static const AVFilterPad compand_inputs[] = {
+    {
+        .name         = "default",
+        .type         = AVMEDIA_TYPE_AUDIO,
+        .filter_frame = filter_frame,
+    },
+    { NULL },
+};
+
+static const AVFilterPad compand_outputs[] = {
+    {
+        .name          = "default",
+        .request_frame = request_frame,
+        .config_props  = config_output,
+        .type          = AVMEDIA_TYPE_AUDIO,
+    },
+    { NULL },
+};
+
+AVFilter avfilter_af_compand = {
+    .name          = "compand",
+    .description   = NULL_IF_CONFIG_SMALL("Compress or expand audio dynamic range."),
+    .query_formats = query_formats,
+    .priv_size     = sizeof(CompandContext),
+    .priv_class    = &compand_class,
+    .init          = init,
+    .uninit        = uninit,
+    .inputs        = compand_inputs,
+    .outputs       = compand_outputs,
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index bda6e3c..bcebcfc 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -80,6 +80,7 @@ void avfilter_register_all(void)
     REGISTER_FILTER(BIQUAD,         biquad,         af);
     REGISTER_FILTER(CHANNELMAP,     channelmap,     af);
     REGISTER_FILTER(CHANNELSPLIT,   channelsplit,   af);
+    REGISTER_FILTER(COMPAND,        compand,        af);
     REGISTER_FILTER(EARWAX,         earwax,         af);
     REGISTER_FILTER(EBUR128,        ebur128,        af);
     REGISTER_FILTER(EQUALIZER,      equalizer,      af);
diff --git a/libavfilter/version.h b/libavfilter/version.h
index 71708bf..190ea2f 100644
--- a/libavfilter/version.h
+++ b/libavfilter/version.h
@@ -30,8 +30,8 @@
 #include "libavutil/avutil.h"
 
 #define LIBAVFILTER_VERSION_MAJOR  3
-#define LIBAVFILTER_VERSION_MINOR  81
-#define LIBAVFILTER_VERSION_MICRO 103
+#define LIBAVFILTER_VERSION_MINOR  82
+#define LIBAVFILTER_VERSION_MICRO 100
 
 #define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \
                                                LIBAVFILTER_VERSION_MINOR, \