avfilter: add asupercut filter

Changelog

@@ -47,6 +47,7 @@ version <next>:
 - DXVA2/D3D11VA hardware accelerated AV1 decoding
 - speechnorm filter
 - SpeedHQ encoder
+- asupercut filter
 
 
 version 4.3:

doc/filters.texi

@@ -1838,7 +1838,7 @@ Set central frequency for band.
 If input doesn't have that frequency the entry is ignored.
 
 @item w
-Set band width in hertz.
+Set band width in Hertz.
 
 @item g
 Set band gain in dB.
@@ -1903,7 +1903,7 @@ Syntax for the commands is : "@var{fN}|f=@var{freq}|w=@var{width}|g=@var{gain}"
 @var{fN} is existing filter number, starting from 0, if no such filter is available
 error is returned.
 @var{freq} set new frequency parameter.
-@var{width} set new width parameter in herz.
+@var{width} set new width parameter in Hertz.
 @var{gain} set new gain parameter in dB.
 
 Full filter invocation with asendcmd may look like this:
@@ -2584,7 +2584,7 @@ Set delay line feedback gain value. Allowed range is from 0 to 1.
 Default value is 0.5.
 
 @item cutoff
-Set cutoff frequency in herz. Allowed range is 50 to 900.
+Set cutoff frequency in Hertz. Allowed range is 50 to 900.
 Default value is 100.
 
 @item slope
@@ -2600,6 +2600,21 @@ Default value is 20.
 
 This filter supports the all above options as @ref{commands}.
 
+@section asupercut
+Cut super frequencies.
+
+The filter accepts the following options:
+
+@table @option
+@item cutoff
+Set cutoff frequency in Hertz. Allowed range is 20000 to 192000.
+Default value is 20000.
+@end table
+
+@subsection Commands
+
+This filter supports the all above options as @ref{commands}.
+
 @section atempo
 
 Adjust audio tempo.

libavfilter/Makefile

@@ -90,6 +90,7 @@ OBJS-$(CONFIG_ASR_FILTER)                    += af_asr.o
 OBJS-$(CONFIG_ASTATS_FILTER)                 += af_astats.o
 OBJS-$(CONFIG_ASTREAMSELECT_FILTER)          += f_streamselect.o framesync.o
 OBJS-$(CONFIG_ASUBBOOST_FILTER)              += af_asubboost.o
+OBJS-$(CONFIG_ASUPERCUT_FILTER)              += af_asupercut.o
 OBJS-$(CONFIG_ATEMPO_FILTER)                 += af_atempo.o
 OBJS-$(CONFIG_ATRIM_FILTER)                  += trim.o
 OBJS-$(CONFIG_AXCORRELATE_FILTER)            += af_axcorrelate.o

libavfilter/af_asupercut.c

@@ -0,0 +1,247 @@
+/*
+ * 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/ffmath.h"
+#include "libavutil/opt.h"
+#include "avfilter.h"
+#include "audio.h"
+#include "formats.h"
+
+typedef struct BiquadCoeffs {
+    double a1, a2;
+    double b0, b1, b2;
+} BiquadCoeffs;
+
+typedef struct ASuperCutContext {
+    const AVClass *class;
+
+    double cutoff;
+
+    int bypass;
+
+    BiquadCoeffs coeffs[5];
+
+    AVFrame *w;
+} ASuperCutContext;
+
+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 get_coeffs(AVFilterContext *ctx)
+{
+    ASuperCutContext *s = ctx->priv;
+    AVFilterLink *inlink = ctx->inputs[0];
+    double w0 = s->cutoff / inlink->sample_rate;
+    double K = tan(M_PI * w0);
+    double q[5];
+
+    s->bypass = w0 >= 0.5;
+    if (s->bypass)
+        return 0;
+
+    q[0] = 0.50623256;
+    q[1] = 0.56116312;
+    q[2] = 0.70710678;
+    q[3] = 1.10134463;
+    q[4] = 3.19622661;
+
+    for (int b = 0; b < 5; b++) {
+        BiquadCoeffs *coeffs = &s->coeffs[b];
+        double norm = 1.0 / (1.0 + K / q[b] + K * K);
+
+        coeffs->b0 = K * K * norm;
+        coeffs->b1 = 2.0 * coeffs->b0;
+        coeffs->b2 = coeffs->b0;
+        coeffs->a1 = -2.0 * (K * K - 1.0) * norm;
+        coeffs->a2 = -(1.0 - K / q[b] + K * K) * norm;
+    }
+
+    return 0;
+}
+
+static int config_input(AVFilterLink *inlink)
+{
+    AVFilterContext *ctx = inlink->dst;
+    ASuperCutContext *s = ctx->priv;
+
+    s->w = ff_get_audio_buffer(inlink, 2 * 5);
+    if (!s->w)
+        return AVERROR(ENOMEM);
+
+    return get_coeffs(ctx);
+}
+
+typedef struct ThreadData {
+    AVFrame *in, *out;
+} ThreadData;
+
+static int filter_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
+{
+    ASuperCutContext *s = ctx->priv;
+    ThreadData *td = arg;
+    AVFrame *out = td->out;
+    AVFrame *in = td->in;
+    const int start = (in->channels * jobnr) / nb_jobs;
+    const int end = (in->channels * (jobnr+1)) / nb_jobs;
+
+    for (int ch = start; ch < end; ch++) {
+        const double *src = (const double *)in->extended_data[ch];
+        double *dst = (double *)out->extended_data[ch];
+
+        for (int b = 0; b < 5; b++) {
+            BiquadCoeffs *coeffs = &s->coeffs[b];
+            const double a1 = coeffs->a1;
+            const double a2 = coeffs->a2;
+            const double b0 = coeffs->b0;
+            const double b1 = coeffs->b1;
+            const double b2 = coeffs->b2;
+            double *w = ((double *)s->w->extended_data[ch]) + b * 2;
+
+            for (int n = 0; n < in->nb_samples; n++) {
+                double sin = b ? dst[n] : src[n];
+                double sout = sin * b0 + w[0];
+
+                w[0] = b1 * sin + w[1] + a1 * sout;
+                w[1] = b2 * sin + a2 * sout;
+
+                dst[n] = sout;
+            }
+        }
+    }
+
+    return 0;
+}
+
+static int filter_frame(AVFilterLink *inlink, AVFrame *in)
+{
+    AVFilterContext *ctx = inlink->dst;
+    ASuperCutContext *s = ctx->priv;
+    AVFilterLink *outlink = ctx->outputs[0];
+    ThreadData td;
+    AVFrame *out;
+
+    if (s->bypass)
+        return ff_filter_frame(outlink, in);
+
+    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);
+    }
+
+    td.in = in; td.out = out;
+    ctx->internal->execute(ctx, filter_channels, &td, NULL, FFMIN(inlink->channels,
+                                                            ff_filter_get_nb_threads(ctx)));
+
+    if (out != in)
+        av_frame_free(&in);
+    return ff_filter_frame(outlink, out);
+}
+
+static int process_command(AVFilterContext *ctx, const char *cmd, const char *args,
+                           char *res, int res_len, int flags)
+{
+    int ret;
+
+    ret = ff_filter_process_command(ctx, cmd, args, res, res_len, flags);
+    if (ret < 0)
+        return ret;
+
+    return get_coeffs(ctx);
+}
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+    ASuperCutContext *s = ctx->priv;
+
+    av_frame_free(&s->w);
+}
+
+#define OFFSET(x) offsetof(ASuperCutContext, x)
+#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
+
+static const AVOption asupercut_options[] = {
+    { "cutoff", "set cutoff frequency", OFFSET(cutoff), AV_OPT_TYPE_DOUBLE, {.dbl=20000}, 20000, 192000, FLAGS },
+    { NULL }
+};
+
+AVFILTER_DEFINE_CLASS(asupercut);
+
+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_asupercut = {
+    .name            = "asupercut",
+    .description     = NULL_IF_CONFIG_SMALL("Cut super frequencies."),
+    .query_formats   = query_formats,
+    .priv_size       = sizeof(ASuperCutContext),
+    .priv_class      = &asupercut_class,
+    .uninit          = uninit,
+    .inputs          = inputs,
+    .outputs         = outputs,
+    .process_command = process_command,
+    .flags           = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC |
+                       AVFILTER_FLAG_SLICE_THREADS,
+};

libavfilter/allfilters.c

@@ -83,6 +83,7 @@ extern AVFilter ff_af_asr;
 extern AVFilter ff_af_astats;
 extern AVFilter ff_af_astreamselect;
 extern AVFilter ff_af_asubboost;
+extern AVFilter ff_af_asupercut;
 extern AVFilter ff_af_atempo;
 extern AVFilter ff_af_atrim;
 extern AVFilter ff_af_axcorrelate;

libavfilter/version.h

@@ -30,7 +30,7 @@
 #include "libavutil/version.h"
 
 #define LIBAVFILTER_VERSION_MAJOR   7
-#define LIBAVFILTER_VERSION_MINOR  90
+#define LIBAVFILTER_VERSION_MINOR  91
 #define LIBAVFILTER_VERSION_MICRO 100