[FFmpeg-devel] [PATCH] avfilter: add adenorm filter
Paul B Mahol
onemda at gmail.com
Mon Nov 2 19:24:52 EET 2020
Signed-off-by: Paul B Mahol <onemda at gmail.com>
---
doc/filters.texi | 27 ++++
libavfilter/Makefile | 1 +
libavfilter/af_adenorm.c | 308 +++++++++++++++++++++++++++++++++++++++
libavfilter/allfilters.c | 1 +
4 files changed, 337 insertions(+)
create mode 100644 libavfilter/af_adenorm.c
diff --git a/doc/filters.texi b/doc/filters.texi
index d98c696f60..777d598c1d 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -731,6 +731,33 @@ adelay=delays=64S:all=1
@end example
@end itemize
+ at section adenorm
+Remedy denormals in audio by adding extremely low-level noise.
+
+A description of the accepted parameters follows.
+
+ at table @option
+ at item level
+Set level of added noise in dB. Default is @code{-351}.
+Allowed range is from -451 to -90.
+
+ at item type
+Set type of added noise.
+
+ at table @option
+ at item dc
+Add DC signal.
+ at item ac
+Add AC signal.
+ at item square
+Add square signal.
+ at item pulse
+Add pulse signal.
+ at end table
+
+ at Default is @code{dc}.
+ at end table
+
@section aderivative, aintegral
Compute derivative/integral of audio stream.
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 1e60c55f6f..028fa50d47 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -41,6 +41,7 @@ OBJS-$(CONFIG_ACUE_FILTER) += f_cue.o
OBJS-$(CONFIG_ADECLICK_FILTER) += af_adeclick.o
OBJS-$(CONFIG_ADECLIP_FILTER) += af_adeclick.o
OBJS-$(CONFIG_ADELAY_FILTER) += af_adelay.o
+OBJS-$(CONFIG_ADENORM_FILTER) += af_adenorm.o
OBJS-$(CONFIG_ADERIVATIVE_FILTER) += af_aderivative.o
OBJS-$(CONFIG_AECHO_FILTER) += af_aecho.o
OBJS-$(CONFIG_AEMPHASIS_FILTER) += af_aemphasis.o
diff --git a/libavfilter/af_adenorm.c b/libavfilter/af_adenorm.c
new file mode 100644
index 0000000000..e689fe556e
--- /dev/null
+++ b/libavfilter/af_adenorm.c
@@ -0,0 +1,308 @@
+/*
+ * 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/avassert.h"
+#include "libavutil/channel_layout.h"
+#include "libavutil/opt.h"
+#include "audio.h"
+#include "avfilter.h"
+#include "internal.h"
+
+enum FilterType {
+ DC_TYPE,
+ AC_TYPE,
+ SQ_TYPE,
+ PS_TYPE,
+ NB_TYPES,
+};
+
+typedef struct ADenormContext {
+ const AVClass *class;
+
+ double level;
+ double level_db;
+ int type;
+ int64_t in_samples;
+
+ void (*filter)(AVFilterContext *ctx, void *dst,
+ const void *src, int nb_samples);
+} ADenormContext;
+
+static int query_formats(AVFilterContext *ctx)
+{
+ AVFilterFormats *formats = NULL;
+ AVFilterChannelLayouts *layouts = NULL;
+ static const enum AVSampleFormat sample_fmts[] = {
+ AV_SAMPLE_FMT_FLTP, 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 void dc_denorm_fltp(AVFilterContext *ctx, void *dstp,
+ const void *srcp, int nb_samples)
+{
+ ADenormContext *s = ctx->priv;
+ const float *src = (const float *)srcp;
+ float *dst = (float *)dstp;
+ const float dc = s->level;
+
+ for (int n = 0; n < nb_samples; n++) {
+ dst[n] = src[n] + dc;
+ }
+}
+
+static void dc_denorm_dblp(AVFilterContext *ctx, void *dstp,
+ const void *srcp, int nb_samples)
+{
+ ADenormContext *s = ctx->priv;
+ const double *src = (const double *)srcp;
+ double *dst = (double *)dstp;
+ const double dc = s->level;
+
+ for (int n = 0; n < nb_samples; n++) {
+ dst[n] = src[n] + dc;
+ }
+}
+
+static void ac_denorm_fltp(AVFilterContext *ctx, void *dstp,
+ const void *srcp, int nb_samples)
+{
+ ADenormContext *s = ctx->priv;
+ const float *src = (const float *)srcp;
+ float *dst = (float *)dstp;
+ const float dc = s->level;
+ const int64_t N = s->in_samples;
+
+ for (int n = 0; n < nb_samples; n++) {
+ dst[n] = src[n] + dc * (((N + n) & 1) ? -1.f : 1.f);
+ }
+}
+
+static void ac_denorm_dblp(AVFilterContext *ctx, void *dstp,
+ const void *srcp, int nb_samples)
+{
+ ADenormContext *s = ctx->priv;
+ const double *src = (const double *)srcp;
+ double *dst = (double *)dstp;
+ const double dc = s->level;
+ const int64_t N = s->in_samples;
+
+ for (int n = 0; n < nb_samples; n++) {
+ dst[n] = src[n] + dc * (((N + n) & 1) ? -1. : 1.);
+ }
+}
+
+static void sq_denorm_fltp(AVFilterContext *ctx, void *dstp,
+ const void *srcp, int nb_samples)
+{
+ ADenormContext *s = ctx->priv;
+ const float *src = (const float *)srcp;
+ float *dst = (float *)dstp;
+ const float dc = s->level;
+ const int64_t N = s->in_samples;
+
+ for (int n = 0; n < nb_samples; n++) {
+ dst[n] = src[n] + dc * ((((N + n) >> 8) & 1) ? -1.f : 1.f);
+ }
+}
+
+static void sq_denorm_dblp(AVFilterContext *ctx, void *dstp,
+ const void *srcp, int nb_samples)
+{
+ ADenormContext *s = ctx->priv;
+ const double *src = (const double *)srcp;
+ double *dst = (double *)dstp;
+ const double dc = s->level;
+ const int64_t N = s->in_samples;
+
+ for (int n = 0; n < nb_samples; n++) {
+ dst[n] = src[n] + dc * ((((N + n) >> 8) & 1) ? -1. : 1.);
+ }
+}
+
+static void ps_denorm_fltp(AVFilterContext *ctx, void *dstp,
+ const void *srcp, int nb_samples)
+{
+ ADenormContext *s = ctx->priv;
+ const float *src = (const float *)srcp;
+ float *dst = (float *)dstp;
+ const float dc = s->level;
+ const int64_t N = s->in_samples;
+
+ for (int n = 0; n < nb_samples; n++) {
+ dst[n] = src[n] + dc * (((N + n) & 255) ? 0.f : 1.f);
+ }
+}
+
+static void ps_denorm_dblp(AVFilterContext *ctx, void *dstp,
+ const void *srcp, int nb_samples)
+{
+ ADenormContext *s = ctx->priv;
+ const double *src = (const double *)srcp;
+ double *dst = (double *)dstp;
+ const double dc = s->level;
+ const int64_t N = s->in_samples;
+
+ for (int n = 0; n < nb_samples; n++) {
+ dst[n] = src[n] + dc * (((N + n) & 255) ? 0. : 1.);
+ }
+}
+
+static int config_output(AVFilterLink *outlink)
+{
+ AVFilterContext *ctx = outlink->src;
+ ADenormContext *s = ctx->priv;
+
+ switch (s->type) {
+ case DC_TYPE:
+ switch (outlink->format) {
+ case AV_SAMPLE_FMT_FLTP: s->filter = dc_denorm_fltp; break;
+ case AV_SAMPLE_FMT_DBLP: s->filter = dc_denorm_dblp; break;
+ }
+ break;
+ case AC_TYPE:
+ switch (outlink->format) {
+ case AV_SAMPLE_FMT_FLTP: s->filter = ac_denorm_fltp; break;
+ case AV_SAMPLE_FMT_DBLP: s->filter = ac_denorm_dblp; break;
+ }
+ break;
+ case SQ_TYPE:
+ switch (outlink->format) {
+ case AV_SAMPLE_FMT_FLTP: s->filter = sq_denorm_fltp; break;
+ case AV_SAMPLE_FMT_DBLP: s->filter = sq_denorm_dblp; break;
+ }
+ break;
+ case PS_TYPE:
+ switch (outlink->format) {
+ case AV_SAMPLE_FMT_FLTP: s->filter = ps_denorm_fltp; break;
+ case AV_SAMPLE_FMT_DBLP: s->filter = ps_denorm_dblp; break;
+ }
+ break;
+ default:
+ av_assert0(0);
+ }
+
+ return 0;
+}
+
+static int filter_frame(AVFilterLink *inlink, AVFrame *in)
+{
+ AVFilterContext *ctx = inlink->dst;
+ ADenormContext *s = ctx->priv;
+ AVFilterLink *outlink = ctx->outputs[0];
+ 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);
+ }
+
+ s->level = exp(s->level_db / 20. * M_LN10);
+ for (int ch = 0; ch < inlink->channels; ch++) {
+ s->filter(ctx, out->extended_data[ch],
+ in->extended_data[ch],
+ in->nb_samples);
+ }
+ s->in_samples += in->nb_samples;
+
+ 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)
+{
+ AVFilterLink *outlink = ctx->outputs[0];
+ int ret;
+
+ ret = ff_filter_process_command(ctx, cmd, args, res, res_len, flags);
+ if (ret < 0)
+ return ret;
+
+ return config_output(outlink);
+}
+
+static const AVFilterPad adenorm_inputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .filter_frame = filter_frame,
+ },
+ { NULL }
+};
+
+static const AVFilterPad adenorm_outputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .config_props = config_output,
+ },
+ { NULL }
+};
+
+#define OFFSET(x) offsetof(ADenormContext, x)
+#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
+
+static const AVOption adenorm_options[] = {
+ { "level", "set level", OFFSET(level_db), AV_OPT_TYPE_DOUBLE, {.dbl=-351}, -451, -90, FLAGS },
+ { "type", "set type", OFFSET(type), AV_OPT_TYPE_INT, {.i64=DC_TYPE}, 0, NB_TYPES-1, FLAGS, "type" },
+ { "dc", NULL, 0, AV_OPT_TYPE_CONST, {.i64=DC_TYPE}, 0, 0, FLAGS, "type"},
+ { "ac", NULL, 0, AV_OPT_TYPE_CONST, {.i64=AC_TYPE}, 0, 0, FLAGS, "type"},
+ { "square",NULL, 0, AV_OPT_TYPE_CONST, {.i64=SQ_TYPE}, 0, 0, FLAGS, "type"},
+ { "pulse", NULL, 0, AV_OPT_TYPE_CONST, {.i64=PS_TYPE}, 0, 0, FLAGS, "type"},
+ { NULL }
+};
+
+AVFILTER_DEFINE_CLASS(adenorm);
+
+AVFilter ff_af_adenorm = {
+ .name = "adenorm",
+ .description = NULL_IF_CONFIG_SMALL("Remedy denormals by adding extremely low-level noise."),
+ .query_formats = query_formats,
+ .priv_size = sizeof(ADenormContext),
+ .inputs = adenorm_inputs,
+ .outputs = adenorm_outputs,
+ .priv_class = &adenorm_class,
+ .process_command = process_command,
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index 801c53f7c0..4c671be329 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -34,6 +34,7 @@ extern AVFilter ff_af_acrusher;
extern AVFilter ff_af_adeclick;
extern AVFilter ff_af_adeclip;
extern AVFilter ff_af_adelay;
+extern AVFilter ff_af_adenorm;
extern AVFilter ff_af_aderivative;
extern AVFilter ff_af_aecho;
extern AVFilter ff_af_aemphasis;
--
2.17.1
More information about the ffmpeg-devel
mailing list