[FFmpeg-cvslog] avfilter: add asubboost filter
Paul B Mahol
git at videolan.org
Thu Apr 30 13:24:11 EEST 2020
ffmpeg | branch: master | Paul B Mahol <onemda at gmail.com> | Wed Apr 22 15:29:16 2020 +0200| [c7d8082357965bfd02bc3a8ba6867b36119d2895] | committer: Paul B Mahol
avfilter: add asubboost filter
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=c7d8082357965bfd02bc3a8ba6867b36119d2895
---
Changelog | 1 +
doc/filters.texi | 39 ++++++++
libavfilter/Makefile | 1 +
libavfilter/af_asubboost.c | 232 +++++++++++++++++++++++++++++++++++++++++++++
libavfilter/allfilters.c | 1 +
libavfilter/version.h | 2 +-
6 files changed, 275 insertions(+), 1 deletion(-)
diff --git a/Changelog b/Changelog
index 83b8a4a46e..9b3e34560f 100644
--- a/Changelog
+++ b/Changelog
@@ -63,6 +63,7 @@ version <next>:
- maskedthreshold filter
- Support for muxing pcm and pgs in m2ts
- Cunning Developments ADPCM decoder
+- asubboost filter
version 4.2:
diff --git a/doc/filters.texi b/doc/filters.texi
index f8dc02f00d..be224bd5b9 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -2454,6 +2454,45 @@ Number of points where the waveform crosses the zero level axis.
Rate of Zero crossings and number of audio samples.
@end table
+ at section asubboost
+Boost subwoofer frequencies.
+
+The filter accepts the following options:
+
+ at table @option
+ at item dry
+Set dry gain, how much of original signal is kept. Allowed range is from 0 to 1.
+Default value is 0.5.
+
+ at item wet
+Set wet gain, how much of filtered signal is kept. Allowed range is from 0 to 1.
+Default value is 0.8.
+
+ at item decay
+Set delay line decay gain value. Allowed range is from 0 to 1.
+Default value is 0.7.
+
+ at item feedback
+Set delay line feedback gain value. Allowed range is from 0 to 1.
+Default value is 0.5.
+
+ at item cutoff
+Set cutoff frequency in herz. Allowed range is 50 to 900.
+Default value is 100.
+
+ at item slope
+Set slope amount for cutoff frequency. Allowed range is 0.0001 to 1.
+Default value is 0.5.
+
+ at item delay
+Set delay. Allowed range is from 1 to 100.
+Default value is 20.
+ 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 e1205eb063..f982afe15f 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -86,6 +86,7 @@ OBJS-$(CONFIG_ASPLIT_FILTER) += split.o
OBJS-$(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_ATEMPO_FILTER) += af_atempo.o
OBJS-$(CONFIG_ATRIM_FILTER) += trim.o
OBJS-$(CONFIG_AXCORRELATE_FILTER) += af_axcorrelate.o
diff --git a/libavfilter/af_asubboost.c b/libavfilter/af_asubboost.c
new file mode 100644
index 0000000000..f8369fd818
--- /dev/null
+++ b/libavfilter/af_asubboost.c
@@ -0,0 +1,232 @@
+/*
+ * 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 ASubBoostContext {
+ const AVClass *class;
+
+ double dry_gain;
+ double wet_gain;
+ double feedback;
+ double decay;
+ double delay;
+ double cutoff;
+ double slope;
+
+ double a0, a1, a2;
+ double b0, b1, b2;
+
+ int write_pos;
+ int buffer_samples;
+
+ AVFrame *i, *o;
+ AVFrame *buffer;
+} ASubBoostContext;
+
+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)
+{
+ ASubBoostContext *s = ctx->priv;
+ AVFilterLink *inlink = ctx->inputs[0];
+ double w0 = 2 * M_PI * s->cutoff / inlink->sample_rate;
+ double alpha = sin(w0) / 2 * sqrt(2. * (1. / s->slope - 1.) + 2.);
+
+ s->a0 = 1 + alpha;
+ s->a1 = -2 * cos(w0);
+ s->a2 = 1 - alpha;
+ s->b0 = (1 - cos(w0)) / 2;
+ s->b1 = 1 - cos(w0);
+ s->b2 = (1 - cos(w0)) / 2;
+
+ s->a1 /= s->a0;
+ s->a2 /= s->a0;
+ s->b0 /= s->a0;
+ s->b1 /= s->a0;
+ s->b2 /= s->a0;
+
+ s->buffer_samples = inlink->sample_rate * s->delay / 1000;
+
+ return 0;
+}
+
+static int config_input(AVFilterLink *inlink)
+{
+ AVFilterContext *ctx = inlink->dst;
+ ASubBoostContext *s = ctx->priv;
+
+ s->buffer = ff_get_audio_buffer(inlink, inlink->sample_rate / 10);
+ s->i = ff_get_audio_buffer(inlink, 2);
+ s->o = ff_get_audio_buffer(inlink, 2);
+ if (!s->buffer || !s->i || !s->o)
+ return AVERROR(ENOMEM);
+
+ return get_coeffs(ctx);
+}
+
+static int filter_frame(AVFilterLink *inlink, AVFrame *in)
+{
+ AVFilterContext *ctx = inlink->dst;
+ AVFilterLink *outlink = ctx->outputs[0];
+ ASubBoostContext *s = ctx->priv;
+ const float wet = s->wet_gain, dry = s->dry_gain, feedback = s->feedback, decay = s->decay;
+ int write_pos;
+ 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);
+ }
+
+ for (int ch = 0; ch < in->channels; ch++) {
+ const double *src = (const double *)in->extended_data[ch];
+ double *dst = (double *)out->extended_data[ch];
+ double *buffer = (double *)s->buffer->extended_data[ch];
+ double *ix = (double *)s->i->extended_data[ch];
+ double *ox = (double *)s->o->extended_data[ch];
+
+ write_pos = s->write_pos;
+ for (int n = 0; n < in->nb_samples; n++) {
+ double out_sample;
+
+ out_sample = src[n] * s->b0 + ix[0] * s->b1 + ix[1] * s->b2 - ox[0] * s->a1 - ox[1] * s->a2;
+ ix[1] = ix[0];
+ ix[0] = src[n];
+ ox[1] = ox[0];
+ ox[0] = out_sample;
+
+ buffer[write_pos] = buffer[write_pos] * decay + out_sample * feedback;
+ dst[n] = src[n] * dry + buffer[write_pos] * wet;
+
+ if (++write_pos >= s->buffer_samples)
+ write_pos = 0;
+ }
+ }
+
+ s->write_pos = write_pos;
+
+ if (out != in)
+ av_frame_free(&in);
+ return ff_filter_frame(outlink, out);
+}
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+ ASubBoostContext *s = ctx->priv;
+
+ av_frame_free(&s->buffer);
+ av_frame_free(&s->i);
+ av_frame_free(&s->o);
+}
+
+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);
+}
+
+#define OFFSET(x) offsetof(ASubBoostContext, x)
+#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
+
+static const AVOption asubboost_options[] = {
+ { "dry", "set dry gain", OFFSET(dry_gain), AV_OPT_TYPE_DOUBLE, {.dbl=0.5}, 0, 1, FLAGS },
+ { "wet", "set wet gain", OFFSET(wet_gain), AV_OPT_TYPE_DOUBLE, {.dbl=0.8}, 0, 1, FLAGS },
+ { "decay", "set decay", OFFSET(decay), AV_OPT_TYPE_DOUBLE, {.dbl=0.7}, 0, 1, FLAGS },
+ { "feedback", "set feedback", OFFSET(feedback), AV_OPT_TYPE_DOUBLE, {.dbl=0.5}, 0, 1, FLAGS },
+ { "cutoff", "set cutoff", OFFSET(cutoff), AV_OPT_TYPE_DOUBLE, {.dbl=100}, 50, 900, FLAGS },
+ { "slope", "set slope", OFFSET(slope), AV_OPT_TYPE_DOUBLE, {.dbl=0.5}, 0.0001, 1, FLAGS },
+ { "delay", "set delay", OFFSET(delay), AV_OPT_TYPE_DOUBLE, {.dbl=20}, 1, 100, FLAGS },
+ { NULL }
+};
+
+AVFILTER_DEFINE_CLASS(asubboost);
+
+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_asubboost = {
+ .name = "asubboost",
+ .description = NULL_IF_CONFIG_SMALL("Boost subwoofer frequencies."),
+ .query_formats = query_formats,
+ .priv_size = sizeof(ASubBoostContext),
+ .priv_class = &asubboost_class,
+ .uninit = uninit,
+ .inputs = inputs,
+ .outputs = outputs,
+ .process_command = process_command,
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index 6354375e91..1b94501da0 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -79,6 +79,7 @@ extern AVFilter ff_af_asplit;
extern AVFilter ff_af_asr;
extern AVFilter ff_af_astats;
extern AVFilter ff_af_astreamselect;
+extern AVFilter ff_af_asubboost;
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 8b06e70615..c0d496684e 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 79
+#define LIBAVFILTER_VERSION_MINOR 80
#define LIBAVFILTER_VERSION_MICRO 100
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