avfilter: add acrossover filter

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

Changelog

@@ -29,6 +29,7 @@ version <next>:
 - AVS2 video encoder via libxavs2
 - amultiply filter
 - Block-Matching 3d (bm3d) denoising filter
+- acrossover filter
 
 
 version 4.0:

doc/filters.texi

@@ -493,6 +493,23 @@ ffmpeg -i first.flac -i second.flac -filter_complex acrossfade=d=10:o=0:c1=exp:c
 @end example
 @end itemize
 
+@section acrossover
+Split audio stream into several bands.
+
+This filter splits audio stream into two or more frequency ranges.
+Summing all streams back will give flat output.
+
+The filter accepts the following options:
+
+@table @option
+@item split
+Set split frequencies. Those must be positive and increasing.
+
+@item order
+Set filter order, can be @var{2nd}, @var{4th} or @var{8th}.
+Default is @var{4th}.
+@end table
+
 @section acrusher
 
 Reduce audio bit resolution.

libavfilter/Makefile

@@ -35,6 +35,7 @@ OBJS-$(CONFIG_ACOMPRESSOR_FILTER)            += af_sidechaincompress.o
 OBJS-$(CONFIG_ACONTRAST_FILTER)              += af_acontrast.o
 OBJS-$(CONFIG_ACOPY_FILTER)                  += af_acopy.o
 OBJS-$(CONFIG_ACROSSFADE_FILTER)             += af_afade.o
+OBJS-$(CONFIG_ACROSSOVER_FILTER)             += af_acrossover.o
 OBJS-$(CONFIG_ACRUSHER_FILTER)               += af_acrusher.o
 OBJS-$(CONFIG_ACUE_FILTER)                   += f_cue.o
 OBJS-$(CONFIG_ADECLICK_FILTER)               += af_adeclick.o

libavfilter/af_acrossover.c

@@ -0,0 +1,343 @@
+/*
+ * 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
+ */
+
+/**
+ * @file
+ * Crossover filter
+ *
+ * Split an audio stream into several bands.
+ */
+
+#include "libavutil/attributes.h"
+#include "libavutil/avstring.h"
+#include "libavutil/channel_layout.h"
+#include "libavutil/internal.h"
+#include "libavutil/opt.h"
+
+#include "audio.h"
+#include "avfilter.h"
+#include "formats.h"
+#include "internal.h"
+
+#define MAX_SPLITS 16
+#define MAX_BANDS MAX_SPLITS + 1
+
+typedef struct BiquadContext {
+    double a0, a1, a2;
+    double b1, b2;
+    double i1, i2;
+    double o1, o2;
+} BiquadContext;
+
+typedef struct CrossoverChannel {
+    BiquadContext lp[MAX_BANDS][4];
+    BiquadContext hp[MAX_BANDS][4];
+} CrossoverChannel;
+
+typedef struct AudioCrossoverContext {
+    const AVClass *class;
+
+    char *splits_str;
+    int order;
+
+    int filter_count;
+    int nb_splits;
+    float *splits;
+
+    CrossoverChannel *xover;
+} AudioCrossoverContext;
+
+#define OFFSET(x) offsetof(AudioCrossoverContext, x)
+#define AF AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_FILTERING_PARAM
+
+static const AVOption acrossover_options[] = {
+    { "split", "set split frequencies", OFFSET(splits_str), AV_OPT_TYPE_STRING, {.str="500"}, 0, 0, AF },
+    { "order", "set order",             OFFSET(order),      AV_OPT_TYPE_INT,    {.i64=1},     0, 2, AF, "m" },
+    { "2nd",   "2nd order",             0,                  AV_OPT_TYPE_CONST,  {.i64=0},     0, 0, AF, "m" },
+    { "4th",   "4th order",             0,                  AV_OPT_TYPE_CONST,  {.i64=1},     0, 0, AF, "m" },
+    { "8th",   "8th order",             0,                  AV_OPT_TYPE_CONST,  {.i64=2},     0, 0, AF, "m" },
+    { NULL }
+};
+
+AVFILTER_DEFINE_CLASS(acrossover);
+
+static av_cold int init(AVFilterContext *ctx)
+{
+    AudioCrossoverContext *s = ctx->priv;
+    char *p, *arg, *saveptr = NULL;
+    int i, ret = 0;
+
+    s->splits = av_calloc(MAX_SPLITS, sizeof(*s->splits));
+    if (!s->splits)
+        return AVERROR(ENOMEM);
+
+    p = s->splits_str;
+    for (i = 0; i < MAX_SPLITS; i++) {
+        float freq;
+
+        if (!(arg = av_strtok(p, " |", &saveptr)))
+            break;
+
+        p = NULL;
+
+        ret = sscanf(arg, "%f", &freq);
+
+        if (freq <= 0) {
+            av_log(ctx, AV_LOG_ERROR, "Frequency %f must be positive number.\n", freq);
+            return AVERROR(EINVAL);
+        }
+
+        if (i > 0 && freq <= s->splits[i-1]) {
+            av_log(ctx, AV_LOG_ERROR, "Frequency %f must be in increasing order.\n", freq);
+            return AVERROR(EINVAL);
+        }
+
+        s->splits[i] = freq;
+    }
+
+    s->nb_splits = i;
+
+    for (i = 0; i <= s->nb_splits; i++) {
+        AVFilterPad pad  = { 0 };
+        char *name;
+
+        pad.type = AVMEDIA_TYPE_AUDIO;
+        name = av_asprintf("out%d", ctx->nb_outputs);
+        if (!name)
+            return AVERROR(ENOMEM);
+        pad.name = name;
+
+        if ((ret = ff_insert_outpad(ctx, i, &pad)) < 0) {
+            av_freep(&pad.name);
+            return ret;
+        }
+    }
+
+    return ret;
+}
+
+static void set_lp(BiquadContext *b, float fc, float q, float sr)
+{
+    double omega = (2.0 * M_PI * fc / sr);
+    double sn = sin(omega);
+    double cs = cos(omega);
+    double alpha = (sn / (2 * q));
+    double inv = (1.0 / (1.0 + alpha));
+
+    b->a2 = b->a0 = (inv * (1.0 - cs) * 0.5);
+    b->a1 = b->a0 + b->a0;
+    b->b1 = -2. * cs * inv;
+    b->b2 = (1. - alpha) * inv;
+}
+
+static void set_hp(BiquadContext *b, float fc, float q, float sr)
+{
+    double omega = 2 * M_PI * fc / sr;
+    double sn = sin(omega);
+    double cs = cos(omega);
+    double alpha = sn / (2 * q);
+    double inv = 1.0 / (1.0 + alpha);
+
+    b->a0 = inv * (1. + cs) / 2.;
+    b->a1 = -2. * b->a0;
+    b->a2 = b->a0;
+    b->b1 = -2. * cs * inv;
+    b->b2 = (1. - alpha) * inv;
+}
+
+static int config_input(AVFilterLink *inlink)
+{
+    AVFilterContext *ctx = inlink->dst;
+    AudioCrossoverContext *s = ctx->priv;
+    int ch, band, sample_rate = inlink->sample_rate;
+    double q;
+
+    s->xover = av_calloc(inlink->channels, sizeof(*s->xover));
+    if (!s->xover)
+        return AVERROR(ENOMEM);
+
+    switch (s->order) {
+    case 0:
+        q = 0.5;
+        s->filter_count = 1;
+        break;
+    case 1:
+        q = M_SQRT1_2;
+        s->filter_count = 2;
+        break;
+    case 2:
+        q = 0.54;
+        s->filter_count = 4;
+        break;
+    }
+
+    for (ch = 0; ch < inlink->channels; ch++) {
+        for (band = 0; band <= s->nb_splits; band++) {
+            set_lp(&s->xover[ch].lp[band][0], s->splits[band], q, sample_rate);
+            set_hp(&s->xover[ch].hp[band][0], s->splits[band], q, sample_rate);
+
+            if (s->order > 1) {
+                set_lp(&s->xover[ch].lp[band][1], s->splits[band], 1.34, sample_rate);
+                set_hp(&s->xover[ch].hp[band][1], s->splits[band], 1.34, sample_rate);
+                set_lp(&s->xover[ch].lp[band][2], s->splits[band],    q, sample_rate);
+                set_hp(&s->xover[ch].hp[band][2], s->splits[band],    q, sample_rate);
+                set_lp(&s->xover[ch].lp[band][3], s->splits[band], 1.34, sample_rate);
+                set_hp(&s->xover[ch].hp[band][3], s->splits[band], 1.34, sample_rate);
+            } else {
+                set_lp(&s->xover[ch].lp[band][1], s->splits[band], q, sample_rate);
+                set_hp(&s->xover[ch].hp[band][1], s->splits[band], q, sample_rate);
+            }
+        }
+    }
+
+    return 0;
+}
+
+static int query_formats(AVFilterContext *ctx)
+{
+    AVFilterFormats *formats;
+    AVFilterChannelLayouts *layouts;
+    static const enum AVSampleFormat sample_fmts[] = {
+        AV_SAMPLE_FMT_DBLP,
+        AV_SAMPLE_FMT_NONE
+    };
+    int 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_make_format_list(sample_fmts);
+    if (!formats)
+        return AVERROR(ENOMEM);
+    ret = ff_set_common_formats(ctx, formats);
+    if (ret < 0)
+        return ret;
+
+    formats = ff_all_samplerates();
+    if (!formats)
+        return AVERROR(ENOMEM);
+    return ff_set_common_samplerates(ctx, formats);
+}
+
+static double biquad_process(BiquadContext *b, double in)
+{
+    double out = in * b->a0 + b->i1 * b->a1 + b->i2 * b->a2 - b->o1 * b->b1 - b->o2 * b->b2;
+
+    b->i2 = b->i1;
+    b->o2 = b->o1;
+    b->i1 = in;
+    b->o1 = out;
+
+    return out;
+}
+
+static int filter_frame(AVFilterLink *inlink, AVFrame *in)
+{
+    AVFilterContext *ctx = inlink->dst;
+    AudioCrossoverContext *s = ctx->priv;
+    AVFrame *frames[MAX_BANDS] = { NULL };
+    int i, f, ch, band, ret = 0;
+
+    for (i = 0; i < ctx->nb_outputs; i++) {
+        frames[i] = ff_get_audio_buffer(ctx->outputs[i], in->nb_samples);
+
+        if (!frames[i]) {
+            ret = AVERROR(ENOMEM);
+            break;
+        }
+
+        frames[i]->pts = in->pts;
+    }
+
+    if (ret < 0)
+        goto fail;
+
+    for (ch = 0; ch < inlink->channels; ch++) {
+        const double *src = (const double *)in->extended_data[ch];
+        CrossoverChannel *xover = &s->xover[ch];
+
+        for (band = 0; band < ctx->nb_outputs; band++) {
+            double *dst = (double *)frames[band]->extended_data[ch];
+
+            for (i = 0; i < in->nb_samples; i++) {
+                dst[i] = src[i];
+
+                for (f = 0; f < s->filter_count; f++) {
+                    if (band + 1 < ctx->nb_outputs) {
+                        BiquadContext *lp = &xover->lp[band][f];
+                        dst[i] = biquad_process(lp, dst[i]);
+                    }
+
+                    if (band - 1 >= 0) {
+                        BiquadContext *hp = &xover->hp[band - 1][f];
+                        dst[i] = biquad_process(hp, dst[i]);
+                    }
+                }
+            }
+        }
+    }
+
+    for (i = 0; i < ctx->nb_outputs; i++) {
+        ret = ff_filter_frame(ctx->outputs[i], frames[i]);
+        if (ret < 0)
+            break;
+    }
+
+fail:
+    av_frame_free(&in);
+
+    return ret;
+}
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+    AudioCrossoverContext *s = ctx->priv;
+    int i;
+
+    av_freep(&s->splits);
+
+    for (i = 0; i < ctx->nb_outputs; i++)
+        av_freep(&ctx->output_pads[i].name);
+}
+
+static const AVFilterPad inputs[] = {
+    {
+        .name         = "default",
+        .type         = AVMEDIA_TYPE_AUDIO,
+        .filter_frame = filter_frame,
+        .config_props = config_input,
+    },
+    { NULL }
+};
+
+AVFilter ff_af_acrossover = {
+    .name           = "acrossover",
+    .description    = NULL_IF_CONFIG_SMALL("Split audio into per-bands streams."),
+    .priv_size      = sizeof(AudioCrossoverContext),
+    .priv_class     = &acrossover_class,
+    .init           = init,
+    .uninit         = uninit,
+    .query_formats  = query_formats,
+    .inputs         = inputs,
+    .outputs        = NULL,
+    .flags          = AVFILTER_FLAG_DYNAMIC_OUTPUTS,
+};

libavfilter/allfilters.c

@@ -29,6 +29,7 @@ extern AVFilter ff_af_acontrast;
 extern AVFilter ff_af_acopy;
 extern AVFilter ff_af_acue;
 extern AVFilter ff_af_acrossfade;
+extern AVFilter ff_af_acrossover;
 extern AVFilter ff_af_acrusher;
 extern AVFilter ff_af_adeclick;
 extern AVFilter ff_af_adeclip;

libavfilter/version.h

@@ -30,7 +30,7 @@
 #include "libavutil/version.h"
 
 #define LIBAVFILTER_VERSION_MAJOR   7
-#define LIBAVFILTER_VERSION_MINOR  31
+#define LIBAVFILTER_VERSION_MINOR  32
 #define LIBAVFILTER_VERSION_MICRO 100
 
 #define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \