@@ -8,6 +8,7 @@ version <next>:

 - Chained Ogg support

 - Theora Midstream reconfiguration support

 - EVRC decoder

+- audio fade filter

 version 1.1:

@@ -282,6 +282,83 @@ aconvert=u8:auto

 @end example

 @end itemize

+@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.

+

+@table @option

+@item type, t

+Specify the effect type, can be either @code{in} for fade-in, or

+@code{out} for a fade-out effect. Default is @code{in}.

+

+@item start_sample, ss

+Specify the number of the start sample for starting to apply the fade

+effect. Default is 0.

+

+@item nb_samples, ns

+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.

+

+@item start_time, st

+Specify time in seconds for starting to apply the fade

+effect. Default is 0.

+If set this option is used instead of @var{start_sample} one.

+

+@item duration, d

+Specify the number of seconds 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 0.

+If set this option is used instead of @var{nb_samples} one.

+

+@item curve

+Set cuve for fade transition.

+@table @option

+@item @var{triangular, linear slope (default)}

+@code{tri}

+@item @var{quarter of sine wave}

+@code{qsin}

+@item @var{half of sine wave}

+@code{esin}

+@item @var{exponential sine wave}

+@code{hsin}

+@item @var{logarithmic}

+@code{log}

+@item @var{inverted parabola}

+@code{par}

+@item @var{quadratic}

+@code{qua}

+@item @var{cubic}

+@code{cub}

+@item @var{square root}

+@code{squ}

+@item @var{cubic root}

+@code{cbr}

+@end table

+@end table

+

+@subsection Examples

+@itemize

+@item

+Fade in first 15 seconds of audio:

+@example

+afade=t=in:ss=0:d=15

+@end example

+

+@item

+Fade out last 25 seconds of a 900 seconds audio:

+@example

+afade=t=out:ss=875:d=25

+@end example

+@end itemize

+

 @section aformat

 Set output format constraints for the input audio. The framework will

@@ -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

new file mode 100644

@@ -0,0 +1,307 @@

+/*

+ * 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;

+    double duration;

+    double start_time;

+

+    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, ESIN, 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 },

+    { "ss",           "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 },

+    { "ns",           "set expression for fade duration in samples", OFFSET(nb_samples),   AV_OPT_TYPE_INT,    {.i64 = 44100}, 1, INT32_MAX, FLAGS },

+    { "start_time",   "set expression of second to start fading",    OFFSET(start_time),   AV_OPT_TYPE_DOUBLE, {.dbl = 0.   }, 0, 7*24*60*60,FLAGS },

+    { "st",           "set expression of second to start fading",    OFFSET(start_time),   AV_OPT_TYPE_DOUBLE, {.dbl = 0.   }, 0, 7*24*60*60,FLAGS },

+    { "duration",     "set expression for fade duration in seconds", OFFSET(duration),     AV_OPT_TYPE_DOUBLE, {.dbl = 0.   }, 0, 24*60*60,  FLAGS },

+    { "d",            "set expression for fade duration in seconds", OFFSET(duration),     AV_OPT_TYPE_DOUBLE, {.dbl = 0.   }, 0, 24*60*60,  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" },

+    { "esin",         "exponential sine wave",                       0,                    AV_OPT_TYPE_CONST,  {.i64 = ESIN }, 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 ESIN:

+        gain = 1.0 - cos(M_PI / 4.0 * (pow(2.0*gain - 1, 3) + 1));

+        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;

+    }

+

+    if (afade->duration)

+        afade->nb_samples = afade->duration * inlink->sample_rate;

+    if (afade->start_time)

+        afade->start_sample = afade->start_time * inlink->sample_rate;

+

+    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,

+};

@@ -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);

@@ -29,8 +29,8 @@

 #include "libavutil/avutil.h"

 #define LIBAVFILTER_VERSION_MAJOR  3

-#define LIBAVFILTER_VERSION_MINOR  32

-#define LIBAVFILTER_VERSION_MICRO 101

+#define LIBAVFILTER_VERSION_MINOR  33

+#define LIBAVFILTER_VERSION_MICRO 100

 #define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \

                                                LIBAVFILTER_VERSION_MINOR, \