[FFmpeg-devel] [PATCH] avfilter: add atilt filter
Paul B Mahol
onemda at gmail.com
Fri Aug 27 00:26:40 EEST 2021
Signed-off-by: Paul B Mahol <onemda at gmail.com>
---
doc/filters.texi | 29 ++++
libavfilter/Makefile | 1 +
libavfilter/af_atilt.c | 287 +++++++++++++++++++++++++++++++++++++++
libavfilter/allfilters.c | 1 +
4 files changed, 318 insertions(+)
create mode 100644 libavfilter/af_atilt.c
diff --git a/doc/filters.texi b/doc/filters.texi
index fb3a683027..f160d70419 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -2989,6 +2989,35 @@ Change filter tempo scale factor.
Syntax for the command is : "@var{tempo}"
@end table
+ at 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:
+
+ at table @option
+ at item freq
+Set central frequency of tilt in Hz. Default is 10000 Hz.
+
+ at item slope
+Set slope direction of tilt. Default is 0. Allowed range is from -1 to 1.
+
+ at item width
+Set width of tilt. Default is 1000. Allowed range is from 100 to 10000.
+
+ at item order
+Set order of tilt filter.
+
+ at item level
+Set input volume level. Allowed range is from 0 to 4.
+Defalt is 1.
+ at end table
+
+ at 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.
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 399a4a5083..2b58325fee 100644
--- a/libavfilter/Makefile
+++ b/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
diff --git a/libavfilter/af_atilt.c b/libavfilter/af_atilt.c
new file mode 100644
index 0000000000..833e0c571b
--- /dev/null
+++ b/libavfilter/af_atilt.c
@@ -0,0 +1,287 @@
+/*
+ * 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,
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index 745fc69e66..6f6677546d 100644
--- a/libavfilter/allfilters.c
+++ b/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;
--
2.17.1
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