Signed-off-by: Paul B Mahol <one...@gmail.com>
---
doc/filters.texi | 15 +++
libavfilter/Makefile | 1 +
libavfilter/af_asupcut.c | 248 +++++++++++++++++++++++++++++++++++++++
libavfilter/allfilters.c | 1 +
4 files changed, 265 insertions(+)
create mode 100644 libavfilter/af_asupcut.c
diff --git a/doc/filters.texi b/doc/filters.texi
index 15acae9709..f8a8012b16 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -2586,6 +2586,21 @@ Default value is 20.
This filter supports the all above options as @ref{commands}.
+@section asupcut
+Cut super frequencies.
+
+The filter accepts the following options:
+
+@table @option
+@item cutoff
+Set cutoff frequency in herz. Allowed range is 20000 to 192000.
+Default value is 20000.
+@end table
+
+@subsection Commands
+
+This filter supports the all above options as @ref{commands}.
+
@section atempo
Adjust audio tempo.
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 36f3d2d0e4..47094b7157 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -90,6 +90,7 @@ OBJS-$(CONFIG_ASR_FILTER) += af_asr.o
OBJS-$(CONFIG_ASTATS_FILTER) += af_astats.o
OBJS-$(CONFIG_ASTREAMSELECT_FILTER) += f_streamselect.o framesync.o
OBJS-$(CONFIG_ASUBBOOST_FILTER) += af_asubboost.o
+OBJS-$(CONFIG_ASUPCUT_FILTER) += af_asupcut.o
OBJS-$(CONFIG_ATEMPO_FILTER) += af_atempo.o
OBJS-$(CONFIG_ATRIM_FILTER) += trim.o
OBJS-$(CONFIG_AXCORRELATE_FILTER) += af_axcorrelate.o
diff --git a/libavfilter/af_asupcut.c b/libavfilter/af_asupcut.c
new file mode 100644
index 0000000000..4a25a12844
--- /dev/null
+++ b/libavfilter/af_asupcut.c
@@ -0,0 +1,248 @@
+/*
+ * 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"
+
+typedef struct BiquadCoeffs {
+ double a1, a2;
+ double b0, b1, b2;
+} BiquadCoeffs;
+
+typedef struct ASupCutContext {
+ const AVClass *class;
+
+ double cutoff;
+
+ int bypass;
+
+ BiquadCoeffs coeffs[5];
+
+ AVFrame *w;
+} ASupCutContext;
+
+static int query_formats(AVFilterContext *ctx)
+{
+ AVFilterFormats *formats = NULL;
+ AVFilterChannelLayouts *layouts = NULL;
+ static const enum AVSampleFormat sample_fmts[] = {
+ AV_SAMPLE_FMT_DBLP,
+ AV_SAMPLE_FMT_NONE
+ };
+ int 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;
+
+ 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_all_samplerates();
+ return ff_set_common_samplerates(ctx, formats);
+}
+
+static int get_coeffs(AVFilterContext *ctx)
+{
+ ASupCutContext *s = ctx->priv;
+ AVFilterLink *inlink = ctx->inputs[0];
+ double w0 = s->cutoff / inlink->sample_rate;
+ double K = tan(M_PI * w0);
+ double q[5];
+
+ if (w0 >= 0.5) {
+ s->bypass = 1;
+ return 0;
+ }
+
+ q[0] = 0.50623256;
+ q[1] = 0.56116312;
+ q[2] = 0.70710678;
+ q[3] = 1.10134463;
+ q[4] = 3.19622661;
+
+ for (int b = 0; b < 5; b++) {
+ BiquadCoeffs *coeffs = &s->coeffs[b];
+ double norm = 1.0 / (1.0 + K / q[b] + K * K);
+
+ coeffs->b0 = K * K * norm;
+ coeffs->b1 = 2.0 * coeffs->b0;
+ coeffs->b2 = coeffs->b0;
+ coeffs->a1 = -2.0 * (K * K - 1.0) * norm;
+ coeffs->a2 = -(1.0 - K / q[b] + K * K) * norm;
+ }
+
+ return 0;
+}
+
+static int config_input(AVFilterLink *inlink)
+{
+ AVFilterContext *ctx = inlink->dst;
+ ASupCutContext *s = ctx->priv;
+
+ s->w = ff_get_audio_buffer(inlink, 2 * 5);
+ if (!s->w)
+ return AVERROR(ENOMEM);
+
+ return get_coeffs(ctx);
+}
+
+typedef struct ThreadData {
+ AVFrame *in, *out;
+} ThreadData;
+
+static int filter_channels(AVFilterContext *ctx, void *arg, int jobnr, int
nb_jobs)
+{
+ ASupCutContext *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++) {
+ const double *src = (const double *)in->extended_data[ch];
+ double *dst = (double *)out->extended_data[ch];
+
+ for (int b = 0; b < 5; b++) {
+ BiquadCoeffs *coeffs = &s->coeffs[b];
+ const double a1 = coeffs->a1;
+ const double a2 = coeffs->a2;
+ const double b0 = coeffs->b0;
+ const double b1 = coeffs->b1;
+ const double b2 = coeffs->b2;
+ double *w = ((double *)s->w->extended_data[ch]) + b * 2;
+
+ for (int n = 0; n < in->nb_samples; n++) {
+ double sin = b ? dst[n] : src[n];
+ double sout = sin * b0 + w[0];
+
+ w[0] = b1 * sin + w[1] + a1 * sout;
+ w[1] = b2 * sin + a2 * sout;
+
+ dst[n] = sout;
+ }
+ }
+ }
+
+ return 0;
+}
+
+static int filter_frame(AVFilterLink *inlink, AVFrame *in)
+{
+ AVFilterContext *ctx = inlink->dst;
+ ASupCutContext *s = ctx->priv;
+ AVFilterLink *outlink = ctx->outputs[0];
+ ThreadData td;
+ AVFrame *out;
+
+ if (s->bypass)
+ return ff_filter_frame(outlink, in);
+
+ 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, 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)
+{
+ ASupCutContext *s = ctx->priv;
+
+ av_frame_free(&s->w);
+}
+
+#define OFFSET(x) offsetof(ASupCutContext, x)
+#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
+
+static const AVOption asupcut_options[] = {
+ { "cutoff", "set cutoff frequency", OFFSET(cutoff), AV_OPT_TYPE_DOUBLE,
{.dbl=20000}, 20000, 192000, FLAGS },
+ { NULL }
+};
+
+AVFILTER_DEFINE_CLASS(asupcut);
+
+static const AVFilterPad inputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .filter_frame = filter_frame,
+ .config_props = config_input,
+ },
+ { NULL }
+};
+
+static const AVFilterPad outputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ },
+ { NULL }
+};
+
+AVFilter ff_af_asupcut = {
+ .name = "asupcut",
+ .description = NULL_IF_CONFIG_SMALL("Cut super frequencies."),
+ .query_formats = query_formats,
+ .priv_size = sizeof(ASupCutContext),
+ .priv_class = &asupcut_class,
+ .uninit = uninit,
+ .inputs = inputs,
+ .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 fde535d50c..bf7fe2ce49 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -83,6 +83,7 @@ extern AVFilter ff_af_asr;
extern AVFilter ff_af_astats;
extern AVFilter ff_af_astreamselect;
extern AVFilter ff_af_asubboost;
+extern AVFilter ff_af_asupcut;
extern AVFilter ff_af_atempo;
extern AVFilter ff_af_atrim;
extern AVFilter ff_af_axcorrelate;
--
2.17.1
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