[FFmpeg-cvslog] avfilter: add adecorrelate filter
Paul B Mahol
git at videolan.org
Sat Aug 28 19:53:21 EEST 2021
ffmpeg | branch: master | Paul B Mahol <onemda at gmail.com> | Mon Aug 23 19:52:32 2021 +0200| [b53a7d2d4d8d45d94d23ff9e31873b82025b1e15] | committer: Paul B Mahol
avfilter: add adecorrelate filter
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=b53a7d2d4d8d45d94d23ff9e31873b82025b1e15
---
Changelog | 1 +
doc/filters.texi | 14 +++
libavfilter/Makefile | 1 +
libavfilter/af_adecorrelate.c | 268 ++++++++++++++++++++++++++++++++++++++++++
libavfilter/allfilters.c | 1 +
libavfilter/version.h | 2 +-
6 files changed, 286 insertions(+), 1 deletion(-)
diff --git a/Changelog b/Changelog
index 20d71bed88..368574fdba 100644
--- a/Changelog
+++ b/Changelog
@@ -12,6 +12,7 @@ version <next>:
- audio and video segment filters
- Apple Graphics (SMC) encoder
- hsvkey and hsvhold video filters
+- adecorrelate audio filter
version 4.4:
diff --git a/doc/filters.texi b/doc/filters.texi
index 44a9a47d1b..c85f82616c 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -745,6 +745,20 @@ Select overlap-save method. Not interpolated samples remain unchanged.
Default value is @code{a}.
@end table
+ at section adecorrelate
+Apply decorrelation to input audio stream.
+
+The filter accepts the following options:
+
+ at table @option
+ at item stages
+Set decorrelation stages of filtering. Allowed
+range is from 1 to 16. Default value is 6.
+
+ at item seed
+Set random seed used for setting delay in samples across channels.
+ at end table
+
@section adelay
Delay one or more audio channels.
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 3166e70b58..399a4a5083 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -40,6 +40,7 @@ OBJS-$(CONFIG_ACRUSHER_FILTER) += af_acrusher.o
OBJS-$(CONFIG_ACUE_FILTER) += f_cue.o
OBJS-$(CONFIG_ADECLICK_FILTER) += af_adeclick.o
OBJS-$(CONFIG_ADECLIP_FILTER) += af_adeclick.o
+OBJS-$(CONFIG_ADECORRELATE_FILTER) += af_adecorrelate.o
OBJS-$(CONFIG_ADELAY_FILTER) += af_adelay.o
OBJS-$(CONFIG_ADENORM_FILTER) += af_adenorm.o
OBJS-$(CONFIG_ADERIVATIVE_FILTER) += af_aderivative.o
diff --git a/libavfilter/af_adecorrelate.c b/libavfilter/af_adecorrelate.c
new file mode 100644
index 0000000000..6113574125
--- /dev/null
+++ b/libavfilter/af_adecorrelate.c
@@ -0,0 +1,268 @@
+/*
+ * Copyright (c) 2013-2020 Michael Barbour <barbour.michael.0 at gmail.com>
+ * Copyright (c) 2021 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/channel_layout.h"
+#include "libavutil/ffmath.h"
+#include "libavutil/lfg.h"
+#include "libavutil/random_seed.h"
+#include "libavutil/opt.h"
+#include "avfilter.h"
+#include "audio.h"
+#include "formats.h"
+
+#define MAX_STAGES 16
+#define FILTER_FC 1100.0
+#define RT60_LF 0.1
+#define RT60_HF 0.008
+
+typedef struct APContext {
+ int len, p;
+ double *mx, *my;
+ double b0, b1, a0, a1;
+} APContext;
+
+typedef struct ADecorrelateContext {
+ const AVClass *class;
+
+ int stages;
+ int64_t seed;
+
+ int nb_channels;
+ APContext (*ap)[MAX_STAGES];
+
+ AVLFG c;
+
+ void (*filter_channel)(AVFilterContext *ctx,
+ int channel,
+ AVFrame *in, AVFrame *out);
+} ADecorrelateContext;
+
+static int query_formats(AVFilterContext *ctx)
+{
+ static const enum AVSampleFormat sample_fmts[] = {
+ AV_SAMPLE_FMT_DBLP,
+ AV_SAMPLE_FMT_NONE
+ };
+ int 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 int ap_init(APContext *ap, int fs, double delay)
+{
+ const int delay_samples = lrint(round(delay * fs));
+ const double gain_lf = -60.0 / (RT60_LF * fs) * delay_samples;
+ const double gain_hf = -60.0 / (RT60_HF * fs) * delay_samples;
+ const double w0 = 2.0 * M_PI * FILTER_FC / fs;
+ const double t = tan(w0 / 2.0);
+ const double g_hf = ff_exp10(gain_hf / 20.0);
+ const double gd = ff_exp10((gain_lf-gain_hf) / 20.0);
+ const double sgd = sqrt(gd);
+
+ ap->len = delay_samples + 1;
+ ap->p = 0;
+ ap->mx = av_calloc(ap->len, sizeof(*ap->mx));
+ ap->my = av_calloc(ap->len, sizeof(*ap->my));
+ if (!ap->mx || !ap->my)
+ return AVERROR(ENOMEM);
+
+ ap->a0 = t + sgd;
+ ap->a1 = (t - sgd) / ap->a0;
+ ap->b0 = (gd*t - sgd) / ap->a0 * g_hf;
+ ap->b1 = (gd*t + sgd) / ap->a0 * g_hf;
+ ap->a0 = 1.0;
+
+ return 0;
+}
+
+static void ap_free(APContext *ap)
+{
+ av_freep(&ap->mx);
+ av_freep(&ap->my);
+}
+
+static double ap_run(APContext *ap, double x)
+{
+ const int i0 = ((ap->p < 1) ? ap->len : ap->p)-1, i_n1 = ap->p, i_n2 = (ap->p+1 >= ap->len) ? 0 : ap->p+1;
+ const double r = ap->b1*x + ap->b0*ap->mx[i0] + ap->a1*ap->mx[i_n2] + ap->a0*ap->mx[i_n1] -
+ ap->a1*ap->my[i0] - ap->b0*ap->my[i_n2] - ap->b1*ap->my[i_n1];
+
+ ap->mx[ap->p] = x;
+ ap->my[ap->p] = r;
+ ap->p = (ap->p+1 >= ap->len) ? 0 : ap->p+1;
+
+ return r;
+}
+
+static void filter_channel_dbl(AVFilterContext *ctx, int ch,
+ AVFrame *in, AVFrame *out)
+{
+ ADecorrelateContext *s = ctx->priv;
+ const double *src = (const double *)in->extended_data[ch];
+ double *dst = (double *)out->extended_data[ch];
+ const int nb_samples = in->nb_samples;
+ const int stages = s->stages;
+ APContext *ap0 = &s->ap[ch][0];
+
+ for (int n = 0; n < nb_samples; n++) {
+ dst[n] = ap_run(ap0, src[n]);
+ for (int i = 1; i < stages; i++) {
+ APContext *ap = &s->ap[ch][i];
+
+ dst[n] = ap_run(ap, dst[n]);
+ }
+ }
+}
+
+static int config_input(AVFilterLink *inlink)
+{
+ AVFilterContext *ctx = inlink->dst;
+ ADecorrelateContext *s = ctx->priv;
+ int ret;
+
+ if (s->seed == -1)
+ s->seed = av_get_random_seed();
+ av_lfg_init(&s->c, s->seed);
+
+ s->nb_channels = inlink->channels;
+ s->ap = av_calloc(inlink->channels, sizeof(*s->ap));
+ if (!s->ap)
+ return AVERROR(ENOMEM);
+
+ for (int i = 0; i < inlink->channels; i++) {
+ for (int j = 0; j < s->stages; j++) {
+ ret = ap_init(&s->ap[i][j], inlink->sample_rate,
+ (double)av_lfg_get(&s->c) / 0xffffffff * 2.2917e-3 + 0.83333e-3);
+ if (ret < 0)
+ return ret;
+ }
+ }
+
+ s->filter_channel = filter_channel_dbl;
+
+ return 0;
+}
+
+typedef struct ThreadData {
+ AVFrame *in, *out;
+} ThreadData;
+
+static int filter_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
+{
+ ADecorrelateContext *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++)
+ s->filter_channel(ctx, ch, in, out);
+
+ return 0;
+}
+
+static int filter_frame(AVFilterLink *inlink, AVFrame *in)
+{
+ AVFilterContext *ctx = inlink->dst;
+ AVFilterLink *outlink = ctx->outputs[0];
+ AVFrame *out;
+ ThreadData td;
+
+ 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;
+ ff_filter_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 av_cold void uninit(AVFilterContext *ctx)
+{
+ ADecorrelateContext *s = ctx->priv;
+
+ if (s->ap) {
+ for (int ch = 0; ch < s->nb_channels; ch++) {
+ for (int stage = 0; stage < s->stages; stage++)
+ ap_free(&s->ap[ch][stage]);
+ }
+ }
+
+ av_freep(&s->ap);
+}
+
+#define OFFSET(x) offsetof(ADecorrelateContext, x)
+#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
+
+static const AVOption adecorrelate_options[] = {
+ { "stages", "set filtering stages", OFFSET(stages), AV_OPT_TYPE_INT, {.i64=6}, 1, MAX_STAGES, FLAGS },
+ { "seed", "set random seed", OFFSET(seed), AV_OPT_TYPE_INT64, {.i64=-1}, -1, UINT_MAX, FLAGS },
+ { NULL }
+};
+
+AVFILTER_DEFINE_CLASS(adecorrelate);
+
+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,
+ },
+};
+
+const AVFilter ff_af_adecorrelate = {
+ .name = "adecorrelate",
+ .description = NULL_IF_CONFIG_SMALL("Apply decorrelation to input audio."),
+ .query_formats = query_formats,
+ .priv_size = sizeof(ADecorrelateContext),
+ .priv_class = &adecorrelate_class,
+ .uninit = uninit,
+ FILTER_INPUTS(inputs),
+ FILTER_OUTPUTS(outputs),
+ .flags = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC |
+ AVFILTER_FLAG_SLICE_THREADS,
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index c49c831203..745fc69e66 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -33,6 +33,7 @@ extern const AVFilter ff_af_acrossover;
extern const AVFilter ff_af_acrusher;
extern const AVFilter ff_af_adeclick;
extern const AVFilter ff_af_adeclip;
+extern const AVFilter ff_af_adecorrelate;
extern const AVFilter ff_af_adelay;
extern const AVFilter ff_af_adenorm;
extern const AVFilter ff_af_aderivative;
diff --git a/libavfilter/version.h b/libavfilter/version.h
index e9a76c5ac3..e3a86d9b01 100644
--- a/libavfilter/version.h
+++ b/libavfilter/version.h
@@ -30,7 +30,7 @@
#include "libavutil/version.h"
#define LIBAVFILTER_VERSION_MAJOR 8
-#define LIBAVFILTER_VERSION_MINOR 4
+#define LIBAVFILTER_VERSION_MINOR 5
#define LIBAVFILTER_VERSION_MICRO 100
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