[FFmpeg-cvslog] avfilter: add adenorm filter

Paul B Mahol git at videolan.org
Thu Nov 5 19:29:29 EET 2020


ffmpeg | branch: master | Paul B Mahol <onemda at gmail.com> | Mon Nov  2 17:16:05 2020 +0100| [a125e081307d1cb3e84eee61fd0485dac317877b] | committer: Paul B Mahol

avfilter: add adenorm filter

> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=a125e081307d1cb3e84eee61fd0485dac317877b
---

 Changelog                |   1 +
 doc/filters.texi         |  27 +++++
 libavfilter/Makefile     |   1 +
 libavfilter/af_adenorm.c | 308 +++++++++++++++++++++++++++++++++++++++++++++++
 libavfilter/allfilters.c |   1 +
 libavfilter/version.h    |   4 +-
 6 files changed, 340 insertions(+), 2 deletions(-)

diff --git a/Changelog b/Changelog
index 3fdcafc355..0ea1901bbe 100644
--- a/Changelog
+++ b/Changelog
@@ -40,6 +40,7 @@ version <next>:
 - High Voltage Software ADPCM encoder
 - LEGO Racers ALP (.tun & .pcm) muxer
 - AV1 VAAPI decoder
+- adenorm filter
 
 
 version 4.3:
diff --git a/doc/filters.texi b/doc/filters.texi
index a6de827ebf..40f8f614fe 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
+
+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 65d03f9191..30e39b7f83 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;
diff --git a/libavfilter/version.h b/libavfilter/version.h
index 44264e12cb..7112ec8942 100644
--- a/libavfilter/version.h
+++ b/libavfilter/version.h
@@ -30,8 +30,8 @@
 #include "libavutil/version.h"
 
 #define LIBAVFILTER_VERSION_MAJOR   7
-#define LIBAVFILTER_VERSION_MINOR  88
-#define LIBAVFILTER_VERSION_MICRO 102
+#define LIBAVFILTER_VERSION_MINOR  89
+#define LIBAVFILTER_VERSION_MICRO 100
 
 
 #define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \



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