avfilter: add atilt filter

Changelog

@@ -13,6 +13,7 @@ version <next>:
 - Apple Graphics (SMC) encoder
 - hsvkey and hsvhold video filters
 - adecorrelate audio filter
+- atilt audio filter
 
 
 version 4.4:

doc/filters.texi

@@ -2989,6 +2989,35 @@ Change filter tempo scale factor.
 Syntax for the command is : "@var{tempo}"
 @end table
 
+@section atilt
+Apply spectral tilt filter to audio stream.
+
+This filter apply any spectral roll-off slope over any specified frequency band.
+
+The filter accepts the following options:
+
+@table @option
+@item freq
+Set central frequency of tilt in Hz. Default is 10000 Hz.
+
+@item slope
+Set slope direction of tilt. Default is 0. Allowed range is from -1 to 1.
+
+@item width
+Set width of tilt. Default is 1000. Allowed range is from 100 to 10000.
+
+@item order
+Set order of tilt filter.
+
+@item level
+Set input volume level. Allowed range is from 0 to 4.
+Defalt is 1.
+@end table
+
+@subsection Commands
+
+This filter supports the all above options as @ref{commands}.
+
 @section atrim
 
 Trim the input so that the output contains one continuous subpart of the input.

libavfilter/Makefile

@@ -99,6 +99,7 @@ OBJS-$(CONFIG_ASUPERCUT_FILTER)              += af_asupercut.o
 OBJS-$(CONFIG_ASUPERPASS_FILTER)             += af_asupercut.o
 OBJS-$(CONFIG_ASUPERSTOP_FILTER)             += af_asupercut.o
 OBJS-$(CONFIG_ATEMPO_FILTER)                 += af_atempo.o
+OBJS-$(CONFIG_ATILT_FILTER)                  += af_atilt.o
 OBJS-$(CONFIG_ATRIM_FILTER)                  += trim.o
 OBJS-$(CONFIG_AXCORRELATE_FILTER)            += af_axcorrelate.o
 OBJS-$(CONFIG_AZMQ_FILTER)                   += f_zmq.o

libavfilter/af_atilt.c

+/*
+ * 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"
+
+#define MAX_ORDER 30
+
+typedef struct Coeffs {
+    double g;
+    double a1;
+    double b0, b1;
+} Coeffs;
+
+typedef struct ATiltContext {
+    const AVClass *class;
+
+    double freq;
+    double level;
+    double slope;
+    double width;
+    int order;
+
+    Coeffs coeffs[MAX_ORDER];
+
+    AVFrame *w;
+
+    int (*filter_channels)(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs);
+} ATiltContext;
+
+static int query_formats(AVFilterContext *ctx)
+{
+    static const enum AVSampleFormat sample_fmts[] = {
+        AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_DBLP,
+        AV_SAMPLE_FMT_NONE
+    };
+    int ret;
+
+    ret = ff_set_common_formats_from_list(ctx, sample_fmts);
+    if (ret < 0)
+        return ret;
+
+    ret = ff_set_common_all_channel_counts(ctx);
+    if (ret < 0)
+        return ret;
+
+    return ff_set_common_all_samplerates(ctx);
+}
+
+static double prewarp(double w, double T, double wp)
+{
+    return wp * tan(w * T * 0.5) / tan(wp * T * 0.5);
+}
+
+static double mz(int i, double w0, double r, double alpha)
+{
+    return w0 * pow(r, -alpha + i);
+}
+
+static double mp(int i, double w0, double r)
+{
+    return w0 * pow(r, i);
+}
+
+static double mzh(int i, double T, double w0, double r, double alpha)
+{
+    return prewarp(mz(i, w0, r, alpha), T, w0);
+}
+
+static double mph(int i, double T, double w0, double r)
+{
+    return prewarp(mp(i, w0, r), T, w0);
+}
+
+static void set_tf1s(Coeffs *coeffs, double b1, double b0, double a0,
+                     double w1, double sr, double alpha)
+{
+    double c = 1.0 / tan(w1 * 0.5 / sr);
+    double d = a0 + c;
+
+    coeffs->b1 = (b0 - b1 * c) / d;
+    coeffs->b0 = (b0 + b1 * c) / d;
+    coeffs->a1 = (a0 - c) / d;
+    coeffs->g = a0 / b0;
+}
+
+static void set_filter(AVFilterContext *ctx,
+                       int order, double sr, double f0,
+                       double bw, double alpha)
+{
+    ATiltContext *s = ctx->priv;
+    const double w0 = 2. * M_PI * f0;
+    const double f1 = f0 + bw;
+    const double w1 = 1.;
+    const double r = pow(f1 / f0, 1.0 / (order - 1.0));
+    const double T = 1. / sr;
+
+    for (int i = 0; i < order; i++) {
+        Coeffs *coeffs = &s->coeffs[i];
+
+        set_tf1s(coeffs, 1.0, mzh(i, T, w0, r, alpha), mph(i, T, w0, r),
+                 w1, sr, alpha);
+    }
+}
+
+static int get_coeffs(AVFilterContext *ctx)
+{
+    ATiltContext *s = ctx->priv;
+    AVFilterLink *inlink = ctx->inputs[0];
+
+    set_filter(ctx, s->order, inlink->sample_rate, s->freq, s->width, s->slope);
+
+    return 0;
+}
+
+typedef struct ThreadData {
+    AVFrame *in, *out;
+} ThreadData;
+
+#define FILTER(name, type)                                          \
+static int filter_channels_## name(AVFilterContext *ctx, void *arg, \
+                                   int jobnr, int nb_jobs)          \
+{                                                                   \
+    ATiltContext *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;           \
+    const type level = s->level;                                    \
+                                                                    \
+    for (int ch = start; ch < end; ch++) {                          \
+        const type *src = (const type *)in->extended_data[ch];      \
+        type *dst = (type *)out->extended_data[ch];                 \
+                                                                    \
+        for (int b = 0; b < s->order; b++) {                        \
+            Coeffs *coeffs = &s->coeffs[b];                         \
+            const type g = coeffs->g;                               \
+            const type a1 = coeffs->a1;                             \
+            const type b0 = coeffs->b0;                             \
+            const type b1 = coeffs->b1;                             \
+            type *w = ((type *)s->w->extended_data[ch]) + b * 2;    \
+                                                                    \
+            for (int n = 0; n < in->nb_samples; n++) {              \
+                type sain = b ? dst[n] : src[n] * level;            \
+                type saout = sain * b0 + w[0] * b1 - w[1] * a1;     \
+                                                                    \
+                w[0] = sain;                                        \
+                w[1] = saout;                                       \
+                                                                    \
+                dst[n] = saout * g;                                 \
+            }                                                       \
+        }                                                           \
+    }                                                               \
+                                                                    \
+    return 0;                                                       \
+}
+
+FILTER(fltp, float)
+FILTER(dblp, double)
+
+static int config_input(AVFilterLink *inlink)
+{
+    AVFilterContext *ctx = inlink->dst;
+    ATiltContext *s = ctx->priv;
+
+    switch (inlink->format) {
+    case AV_SAMPLE_FMT_FLTP: s->filter_channels = filter_channels_fltp; break;
+    case AV_SAMPLE_FMT_DBLP: s->filter_channels = filter_channels_dblp; break;
+    }
+
+    s->w = ff_get_audio_buffer(inlink, 2 * MAX_ORDER);
+    if (!s->w)
+        return AVERROR(ENOMEM);
+
+    return get_coeffs(ctx);
+}
+
+static int filter_frame(AVFilterLink *inlink, AVFrame *in)
+{
+    AVFilterContext *ctx = inlink->dst;
+    ATiltContext *s = ctx->priv;
+    AVFilterLink *outlink = ctx->outputs[0];
+    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);
+    }
+
+    td.in = in; td.out = out;
+    ctx->internal->execute(ctx, s->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)
+{
+    ATiltContext *s = ctx->priv;
+
+    av_frame_free(&s->w);
+}
+
+#define OFFSET(x) offsetof(ATiltContext, x)
+#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
+
+static const AVOption atilt_options[] = {
+    { "freq",   "set central frequency",OFFSET(freq),   AV_OPT_TYPE_DOUBLE, {.dbl=10000},    20, 192000, FLAGS },
+    { "slope",  "set filter slope",     OFFSET(slope),  AV_OPT_TYPE_DOUBLE, {.dbl=0},        -1,      1, FLAGS },
+    { "width",  "set filter width",     OFFSET(width),  AV_OPT_TYPE_DOUBLE, {.dbl=1000},    100,  10000, FLAGS },
+    { "order",  "set filter order",     OFFSET(order),  AV_OPT_TYPE_INT,    {.i64=5},       2,MAX_ORDER, FLAGS },
+    { "level",  "set input level",      OFFSET(level),  AV_OPT_TYPE_DOUBLE, {.dbl=1.},        0.,    4., FLAGS },
+    { NULL }
+};
+
+AVFILTER_DEFINE_CLASS(atilt);
+
+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,
+    },
+};
+
+AVFilter ff_af_atilt = {
+    .name            = "atilt",
+    .description     = NULL_IF_CONFIG_SMALL("Apply spectral tilt to audio."),
+    .query_formats   = query_formats,
+    .priv_size       = sizeof(ATiltContext),
+    .priv_class      = &atilt_class,
+    .uninit          = uninit,
+    FILTER_INPUTS(inputs),
+    FILTER_OUTPUTS(outputs),
+    .process_command = process_command,
+    .flags           = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC |
+                       AVFILTER_FLAG_SLICE_THREADS,
+};

libavfilter/allfilters.c

@@ -92,6 +92,7 @@ extern const AVFilter ff_af_asupercut;
 extern const AVFilter ff_af_asuperpass;
 extern const AVFilter ff_af_asuperstop;
 extern const AVFilter ff_af_atempo;
+extern const AVFilter ff_af_atilt;
 extern const AVFilter ff_af_atrim;
 extern const AVFilter ff_af_axcorrelate;
 extern const AVFilter ff_af_azmq;

libavfilter/version.h

@@ -30,7 +30,7 @@
 #include "libavutil/version.h"
 
 #define LIBAVFILTER_VERSION_MAJOR   8
-#define LIBAVFILTER_VERSION_MINOR   5
+#define LIBAVFILTER_VERSION_MINOR   6
 #define LIBAVFILTER_VERSION_MICRO 100