[FFmpeg-cvslog] avfilter: add asupercut filter
Paul B Mahol
git at videolan.org
Thu Nov 26 18:41:01 EET 2020
ffmpeg | branch: master | Paul B Mahol <onemda at gmail.com> | Mon Nov 23 18:45:54 2020 +0100| [3c922681c35ac6f58e4a4bc02b8f0966b308d985] | committer: Paul B Mahol
avfilter: add asupercut filter
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=3c922681c35ac6f58e4a4bc02b8f0966b308d985
---
Changelog | 1 +
doc/filters.texi | 21 +++-
libavfilter/Makefile | 1 +
libavfilter/af_asupercut.c | 247 +++++++++++++++++++++++++++++++++++++++++++++
libavfilter/allfilters.c | 1 +
libavfilter/version.h | 2 +-
6 files changed, 269 insertions(+), 4 deletions(-)
diff --git a/Changelog b/Changelog
index 8b2e17378c..ebb1727875 100644
--- a/Changelog
+++ b/Changelog
@@ -47,6 +47,7 @@ version <next>:
- DXVA2/D3D11VA hardware accelerated AV1 decoding
- speechnorm filter
- SpeedHQ encoder
+- asupercut filter
version 4.3:
diff --git a/doc/filters.texi b/doc/filters.texi
index 109cacc23d..488d3c4654 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -1838,7 +1838,7 @@ Set central frequency for band.
If input doesn't have that frequency the entry is ignored.
@item w
-Set band width in hertz.
+Set band width in Hertz.
@item g
Set band gain in dB.
@@ -1903,7 +1903,7 @@ Syntax for the commands is : "@var{fN}|f=@var{freq}|w=@var{width}|g=@var{gain}"
@var{fN} is existing filter number, starting from 0, if no such filter is available
error is returned.
@var{freq} set new frequency parameter.
- at var{width} set new width parameter in herz.
+ at var{width} set new width parameter in Hertz.
@var{gain} set new gain parameter in dB.
Full filter invocation with asendcmd may look like this:
@@ -2584,7 +2584,7 @@ Set delay line feedback gain value. Allowed range is from 0 to 1.
Default value is 0.5.
@item cutoff
-Set cutoff frequency in herz. Allowed range is 50 to 900.
+Set cutoff frequency in Hertz. Allowed range is 50 to 900.
Default value is 100.
@item slope
@@ -2600,6 +2600,21 @@ Default value is 20.
This filter supports the all above options as @ref{commands}.
+ at section asupercut
+Cut super frequencies.
+
+The filter accepts the following options:
+
+ at table @option
+ at item cutoff
+Set cutoff frequency in Hertz. Allowed range is 20000 to 192000.
+Default value is 20000.
+ at end table
+
+ at 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 794a55ac3d..cff9402989 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-$(CONFIG_ASUPERCUT_FILTER) += af_asupercut.o
OBJS-$(CONFIG_ATEMPO_FILTER) += af_atempo.o
OBJS-$(CONFIG_ATRIM_FILTER) += trim.o
OBJS-$(CONFIG_AXCORRELATE_FILTER) += af_axcorrelate.o
diff --git a/libavfilter/af_asupercut.c b/libavfilter/af_asupercut.c
new file mode 100644
index 0000000000..a22830c2e8
--- /dev/null
+++ b/libavfilter/af_asupercut.c
@@ -0,0 +1,247 @@
+/*
+ * 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 ASuperCutContext {
+ const AVClass *class;
+
+ double cutoff;
+
+ int bypass;
+
+ BiquadCoeffs coeffs[5];
+
+ AVFrame *w;
+} ASuperCutContext;
+
+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)
+{
+ ASuperCutContext *s = ctx->priv;
+ AVFilterLink *inlink = ctx->inputs[0];
+ double w0 = s->cutoff / inlink->sample_rate;
+ double K = tan(M_PI * w0);
+ double q[5];
+
+ s->bypass = w0 >= 0.5;
+ if (s->bypass)
+ 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;
+ ASuperCutContext *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)
+{
+ ASuperCutContext *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;
+ ASuperCutContext *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)
+{
+ ASuperCutContext *s = ctx->priv;
+
+ av_frame_free(&s->w);
+}
+
+#define OFFSET(x) offsetof(ASuperCutContext, x)
+#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
+
+static const AVOption asupercut_options[] = {
+ { "cutoff", "set cutoff frequency", OFFSET(cutoff), AV_OPT_TYPE_DOUBLE, {.dbl=20000}, 20000, 192000, FLAGS },
+ { NULL }
+};
+
+AVFILTER_DEFINE_CLASS(asupercut);
+
+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_asupercut = {
+ .name = "asupercut",
+ .description = NULL_IF_CONFIG_SMALL("Cut super frequencies."),
+ .query_formats = query_formats,
+ .priv_size = sizeof(ASuperCutContext),
+ .priv_class = &asupercut_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 fbfd8989c6..83f434bc27 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_asupercut;
extern AVFilter ff_af_atempo;
extern AVFilter ff_af_atrim;
extern AVFilter ff_af_axcorrelate;
diff --git a/libavfilter/version.h b/libavfilter/version.h
index 23c9d374ad..bd20eaee73 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 90
+#define LIBAVFILTER_VERSION_MINOR 91
#define LIBAVFILTER_VERSION_MICRO 100
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