[FFmpeg-devel] [PATCH] afade filter

Paul B Mahol onemda at gmail.com
Sun Jan 20 18:51:01 CET 2013


Signed-off-by: Paul B Mahol <onemda at gmail.com>
---
 doc/filters.texi         |  63 ++++++++++
 libavfilter/Makefile     |   1 +
 libavfilter/af_afade.c   | 292 +++++++++++++++++++++++++++++++++++++++++++++++
 libavfilter/allfilters.c |   1 +
 4 files changed, 357 insertions(+)
 create mode 100644 libavfilter/af_afade.c

diff --git a/doc/filters.texi b/doc/filters.texi
index 42c78b8..806ff48 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -282,6 +282,69 @@ aconvert=u8:auto
 @end example
 @end itemize
 
+ at section afade
+
+Apply fade-in/out effect to input audio.
+
+The filter accepts parameters as a list of @var{key}=@var{value}
+pairs, separated by ":".
+
+A description of the accepted parameters follows.
+
+ at table @option
+ at item type, t
+Specify the effect type, can be either @code{in} for fade-in, or
+ at code{out} for a fade-out effect. Default is @code{in}.
+
+ at item start_sample, s
+Specify the number of the start sample for starting to apply the fade
+effect. Default is 0.
+
+ at item nb_frames, n
+Specify the number of samples for which the fade effect has to last. At
+the end of the fade-in effect the output audio will have the same
+volume as the input audio, at the end of the fade-out transition
+the output audio will be silence. Default is 44100.
+
+ at item curve
+Set cuve for fade transition.
+ at table @option
+ at item @var{triangular, linear slope}
+ at code{tri}
+ at item @var{quarter of sine wave}
+ at code{qsin}
+ at item @var{half of sine wave}
+ at code{hsin}
+ at item @var{logarithmic}
+ at code{log}
+ at item @var{inverted parabola}
+ at code{par}
+ at item @var{quadratic}
+ at code{qua}
+ at item @var{cubic}
+ at code{cub}
+ at item @var{square root}
+ at code{squ}
+ at item @var{cubic root}
+ at code{cbr}
+ at end table
+ at end table
+
+ at subsection Examples
+ at itemize
+ at item
+Fade in first 15 seconds of audio with 44100 sample rate:
+ at example
+afade=t=in:s=0:n=661500
+ at end example
+
+ at item
+Fade out last 25 seconds of a 900 seconds audio with 44100 sample rate:
+ at example
+afade=t=out:s=38587500:n=102500
+ at end example
+ at end itemize
+
 @section aformat
 
 Set output format constraints for the input audio. The framework will
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 9bb9fa3..5835a7e 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -51,6 +51,7 @@ OBJS-$(CONFIG_AVFORMAT)                      += lavfutils.o
 OBJS-$(CONFIG_SWSCALE)                       += lswsutils.o
 
 OBJS-$(CONFIG_ACONVERT_FILTER)               += af_aconvert.o
+OBJS-$(CONFIG_AFADE_FILTER)                  += af_afade.o
 OBJS-$(CONFIG_AFORMAT_FILTER)                += af_aformat.o
 OBJS-$(CONFIG_AMERGE_FILTER)                 += af_amerge.o
 OBJS-$(CONFIG_AMIX_FILTER)                   += af_amix.o
diff --git a/libavfilter/af_afade.c b/libavfilter/af_afade.c
new file mode 100644
index 0000000..4f14b7c
--- /dev/null
+++ b/libavfilter/af_afade.c
@@ -0,0 +1,292 @@
+/*
+ * Copyright (c) 2013 Paul B Mahol
+ *
+ * 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
+ */
+
+/**
+ * @file
+ * fade audio filter
+ */
+
+#include "libavutil/opt.h"
+#include "audio.h"
+#include "avfilter.h"
+#include "internal.h"
+
+typedef struct {
+    const AVClass *class;
+    int type;
+    int curve;
+    int nb_samples;
+    int64_t start_sample;
+
+    void (*fade_samples)(uint8_t **dst, uint8_t * const *src,
+                         int nb_samples, int channels, int direction,
+                         int64_t start, int range, int curve);
+} AudioFadeContext;
+
+enum CurveType { TRI, QSIN, HSIN, LOG, PAR, QUA, CUB, SQU, CBR };
+
+#define OFFSET(x) offsetof(AudioFadeContext, x)
+#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
+
+static const AVOption afade_options[] = {
+    { "type",         "set the fade direction",                      OFFSET(type),         AV_OPT_TYPE_INT,    {.i64 = 0    }, 0, 1, FLAGS, "type" },
+    { "t",            "set the fade direction",                      OFFSET(type),         AV_OPT_TYPE_INT,    {.i64 = 0    }, 0, 1, FLAGS, "type" },
+    { "in",           NULL,                                          0,                    AV_OPT_TYPE_CONST,  {.i64 = 0    }, 0, 0, FLAGS, "type" },
+    { "out",          NULL,                                          0,                    AV_OPT_TYPE_CONST,  {.i64 = 1    }, 0, 0, FLAGS, "type" },
+    { "start_sample", "set expression of sample to start fading",    OFFSET(start_sample), AV_OPT_TYPE_INT64,  {.i64 = 0    }, 0, INT64_MAX, FLAGS },
+    { "st",           "set expression of sample to start fading",    OFFSET(start_sample), AV_OPT_TYPE_INT64,  {.i64 = 0    }, 0, INT64_MAX, FLAGS },
+    { "nb_samples",   "set expression for fade duration in samples", OFFSET(nb_samples),   AV_OPT_TYPE_INT,    {.i64 = 44100}, 1, INT32_MAX, FLAGS },
+    { "n",            "set expression for fade duration in samples", OFFSET(nb_samples),   AV_OPT_TYPE_INT,    {.i64 = 44100}, 1, INT32_MAX, FLAGS },
+    { "curve",        "set expression for fade curve",               OFFSET(curve),        AV_OPT_TYPE_INT,    {.i64 = TRI  }, TRI, CBR, FLAGS, "curve" },
+    { "c",            "set expression for fade curve",               OFFSET(curve),        AV_OPT_TYPE_INT,    {.i64 = TRI  }, TRI, CBR, FLAGS, "curve" },
+    { "tri",          "linear slope",                                0,                    AV_OPT_TYPE_CONST,  {.i64 = TRI  }, 0, 0, FLAGS, "curve" },
+    { "qsin",         "quarter of sine wave",                        0,                    AV_OPT_TYPE_CONST,  {.i64 = QSIN }, 0, 0, FLAGS, "curve" },
+    { "hsin",         "half of sine wave",                           0,                    AV_OPT_TYPE_CONST,  {.i64 = HSIN }, 0, 0, FLAGS, "curve" },
+    { "log",          "logarithmic",                                 0,                    AV_OPT_TYPE_CONST,  {.i64 = LOG  }, 0, 0, FLAGS, "curve" },
+    { "par",          "inverted parabola",                           0,                    AV_OPT_TYPE_CONST,  {.i64 = PAR  }, 0, 0, FLAGS, "curve" },
+    { "qua",          "quadratic",                                   0,                    AV_OPT_TYPE_CONST,  {.i64 = QUA  }, 0, 0, FLAGS, "curve" },
+    { "cub",          "cubic",                                       0,                    AV_OPT_TYPE_CONST,  {.i64 = CUB  }, 0, 0, FLAGS, "curve" },
+    { "squ",          "square root",                                 0,                    AV_OPT_TYPE_CONST,  {.i64 = SQU  }, 0, 0, FLAGS, "curve" },
+    { "cbr",          "cubic root",                                  0,                    AV_OPT_TYPE_CONST,  {.i64 = CBR  }, 0, 0, FLAGS, "curve" },
+    {NULL},
+};
+
+AVFILTER_DEFINE_CLASS(afade);
+
+static av_cold int init(AVFilterContext *ctx, const char *args)
+{
+    AudioFadeContext *afade = ctx->priv;
+    int ret;
+
+    afade->class = &afade_class;
+    av_opt_set_defaults(afade);
+
+    if ((ret = av_set_options_string(afade, args, "=", ":")) < 0)
+        return ret;
+
+    if (INT64_MAX - afade->nb_samples < afade->start_sample)
+        return AVERROR(EINVAL);
+
+    return 0;
+}
+
+static int query_formats(AVFilterContext *ctx)
+{
+    AVFilterFormats *formats;
+    AVFilterChannelLayouts *layouts;
+    static const enum AVSampleFormat sample_fmts[] = {
+        AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16P,
+        AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32P,
+        AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP,
+        AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBLP,
+        AV_SAMPLE_FMT_NONE
+    };
+
+    layouts = ff_all_channel_layouts();
+    if (!layouts)
+        return AVERROR(ENOMEM);
+    ff_set_common_channel_layouts(ctx, layouts);
+
+    formats = ff_make_format_list(sample_fmts);
+    if (!formats)
+        return AVERROR(ENOMEM);
+    ff_set_common_formats(ctx, formats);
+
+    formats = ff_all_samplerates();
+    if (!formats)
+        return AVERROR(ENOMEM);
+    ff_set_common_samplerates(ctx, formats);
+
+    return 0;
+}
+
+static double fade_gain(int curve, int64_t index, int range)
+{
+    double gain;
+
+    gain = FFMAX(0.0, FFMIN(1.0, 1.0 * index / range));
+
+    switch (curve) {
+    case QSIN:
+        gain = sin(gain * M_PI / 2.0);
+        break;
+    case HSIN:
+        gain = (1.0 - cos(gain * M_PI)) / 2.0;
+        break;
+    case LOG:
+        gain = pow(0.1, (1 - gain) * 5.0);
+        break;
+    case PAR:
+        gain = (1 - (1 - gain) * (1 - gain));
+        break;
+    case QUA:
+        gain *= gain;
+        break;
+    case CUB:
+        gain = gain * gain * gain;
+        break;
+    case SQU:
+        gain = sqrt(gain);
+        break;
+    case CBR:
+        gain = cbrt(gain);
+        break;
+    }
+
+    return gain;
+}
+
+#define FADE_PLANAR(name, type)                                             \
+static void fade_samples_## name ##p(uint8_t **dst, uint8_t * const *src,   \
+                                     int nb_samples, int channels, int dir, \
+                                     int64_t start, int range, int curve)   \
+{                                                                           \
+    int i, c;                                                               \
+                                                                            \
+    for (i = 0; i < nb_samples; i++) {                                      \
+        double gain = fade_gain(curve, start + i * dir, range);             \
+        for (c = 0; c < channels; c++) {                                    \
+            type *d = (type *)dst[c];                                       \
+            const type *s = (type *)src[c];                                 \
+                                                                            \
+            d[i] = s[i] * gain;                                             \
+        }                                                                   \
+    }                                                                       \
+}
+
+#define FADE(name, type)                                                    \
+static void fade_samples_## name (uint8_t **dst, uint8_t * const *src,      \
+                                  int nb_samples, int channels, int dir,    \
+                                  int64_t start, int range, int curve)      \
+{                                                                           \
+    type *d = (type *)dst[0];                                               \
+    const type *s = (type *)src[0];                                         \
+    int i, c, k = 0;                                                        \
+                                                                            \
+    for (i = 0; i < nb_samples; i++) {                                      \
+        double gain = fade_gain(curve, start + i * dir, range);             \
+        for (c = 0; c < channels; c++, k++)                                 \
+            d[k] = s[k] * gain;                                             \
+    }                                                                       \
+}
+
+FADE_PLANAR(dbl, double)
+FADE_PLANAR(flt, float)
+FADE_PLANAR(s16, int16_t)
+FADE_PLANAR(s32, int32_t)
+
+FADE(dbl, double)
+FADE(flt, float)
+FADE(s16, int16_t)
+FADE(s32, int32_t)
+
+static int config_output(AVFilterLink *outlink)
+{
+    AVFilterContext *ctx    = outlink->src;
+    AudioFadeContext *afade = ctx->priv;
+    AVFilterLink *inlink    = ctx->inputs[0];
+
+    switch (inlink->format) {
+    case AV_SAMPLE_FMT_DBL:  afade->fade_samples = fade_samples_dbl;  break;
+    case AV_SAMPLE_FMT_DBLP: afade->fade_samples = fade_samples_dblp; break;
+    case AV_SAMPLE_FMT_FLT:  afade->fade_samples = fade_samples_flt;  break;
+    case AV_SAMPLE_FMT_FLTP: afade->fade_samples = fade_samples_fltp; break;
+    case AV_SAMPLE_FMT_S16:  afade->fade_samples = fade_samples_s16;  break;
+    case AV_SAMPLE_FMT_S16P: afade->fade_samples = fade_samples_s16p; break;
+    case AV_SAMPLE_FMT_S32:  afade->fade_samples = fade_samples_s32;  break;
+    case AV_SAMPLE_FMT_S32P: afade->fade_samples = fade_samples_s32p; break;
+    }
+
+    return 0;
+}
+
+static int filter_frame(AVFilterLink *inlink, AVFilterBufferRef *buf)
+{
+    AudioFadeContext *afade = inlink->dst->priv;
+    AVFilterLink *outlink   = inlink->dst->outputs[0];
+    int nb_samples          = buf->audio->nb_samples;
+    AVFilterBufferRef *out_buf;
+    int64_t cur_sample = av_rescale_q(buf->pts, (AVRational){1, outlink->sample_rate}, outlink->time_base);
+
+    if ((!afade->type && (afade->start_sample + afade->nb_samples < cur_sample)) ||
+        ( afade->type && (cur_sample + afade->nb_samples < afade->start_sample)))
+        return ff_filter_frame(outlink, buf);
+
+    if (buf->perms & AV_PERM_WRITE) {
+        out_buf = buf;
+    } else {
+        out_buf = ff_get_audio_buffer(inlink, AV_PERM_WRITE, nb_samples);
+        if (!out_buf)
+            return AVERROR(ENOMEM);
+        out_buf->pts = buf->pts;
+    }
+
+    if ((!afade->type && (cur_sample + nb_samples < afade->start_sample)) ||
+        ( afade->type && (afade->start_sample + afade->nb_samples < cur_sample))) {
+        av_samples_set_silence(out_buf->extended_data, 0, nb_samples,
+                               out_buf->audio->channels, out_buf->format);
+    } else {
+        int64_t start;
+
+        if (!afade->type)
+            start = cur_sample - afade->start_sample;
+        else
+            start = afade->start_sample + afade->nb_samples - cur_sample;
+
+        afade->fade_samples(out_buf->extended_data, buf->extended_data,
+                            nb_samples, buf->audio->channels,
+                            afade->type ? -1 : 1, start,
+                            afade->nb_samples, afade->curve);
+    }
+
+    if (buf != out_buf)
+        avfilter_unref_buffer(buf);
+
+    return ff_filter_frame(outlink, out_buf);
+}
+
+static const AVFilterPad avfilter_af_afade_inputs[] = {
+    {
+        .name         = "default",
+        .type         = AVMEDIA_TYPE_AUDIO,
+        .filter_frame = filter_frame,
+    },
+    { NULL }
+};
+
+static const AVFilterPad avfilter_af_afade_outputs[] = {
+    {
+        .name         = "default",
+        .type         = AVMEDIA_TYPE_AUDIO,
+        .config_props = config_output,
+    },
+    { NULL }
+};
+
+AVFilter avfilter_af_afade = {
+    .name          = "afade",
+    .description   = NULL_IF_CONFIG_SMALL("Fade in/out input audio."),
+    .query_formats = query_formats,
+    .priv_size     = sizeof(AudioFadeContext),
+    .init          = init,
+    .inputs        = avfilter_af_afade_inputs,
+    .outputs       = avfilter_af_afade_outputs,
+    .priv_class    = &afade_class,
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index 4815c4a..24df561 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -45,6 +45,7 @@ void avfilter_register_all(void)
     initialized = 1;
 
     REGISTER_FILTER(ACONVERT,       aconvert,       af);
+    REGISTER_FILTER(AFADE,          afade,          af);
     REGISTER_FILTER(AFORMAT,        aformat,        af);
     REGISTER_FILTER(AMERGE,         amerge,         af);
     REGISTER_FILTER(AMIX,           amix,           af);
-- 
1.7.11.4



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