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-rw-r--r--Changelog1
-rw-r--r--doc/filters.texi17
-rw-r--r--libavfilter/Makefile1
-rw-r--r--libavfilter/af_acrossover.c343
-rw-r--r--libavfilter/allfilters.c1
-rw-r--r--libavfilter/version.h2
6 files changed, 364 insertions, 1 deletions
diff --git a/Changelog b/Changelog
index 69d70a9ce6..8b839360e7 100644
--- a/Changelog
+++ b/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:
diff --git a/doc/filters.texi b/doc/filters.texi
index 20e0a3ec63..5cc96bc1cc 100644
--- a/doc/filters.texi
+++ b/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.
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 190ce2861c..67e20cc858 100644
--- a/libavfilter/Makefile
+++ b/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
diff --git a/libavfilter/af_acrossover.c b/libavfilter/af_acrossover.c
new file mode 100644
index 0000000000..9acf3f14e4
--- /dev/null
+++ b/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,
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index 6a3bb2fc33..8b1c0d618c 100644
--- a/libavfilter/allfilters.c
+++ b/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;
diff --git a/libavfilter/version.h b/libavfilter/version.h
index 25d5694027..25ec7f58b4 100644
--- a/libavfilter/version.h
+++ b/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, \
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