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afade filter

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Signed-off-by: Paul B Mahol <onemda@gmail.com>
This commit is contained in:
Paul B Mahol 2013-01-17 21:53:27 +00:00
parent 5f61e09a8f
commit 6ea8a830e8
6 changed files with 389 additions and 2 deletions
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1
Changelog
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@ -8,6 +8,7 @@ version <next>:
- Chained Ogg support
- Theora Midstream reconfiguration support
- EVRC decoder
- audio fade filter
version 1.1:

77
doc/filters.texi
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@ -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

1
libavfilter/Makefile
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@ -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

307
libavfilter/af_afade.c Normal file
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@ -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,
};

1
libavfilter/allfilters.c
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@ -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);

4
libavfilter/version.h
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@ -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, \