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afade filter
Signed-off-by: Paul B Mahol <onemda@gmail.com>
This commit is contained in:
Paul B Mahol
parent
5f61e09a8f
commit
6ea8a830e8
@ -8,6 +8,7 @@ version <next>:
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- Chained Ogg support
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- Chained Ogg support
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- Theora Midstream reconfiguration support
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- Theora Midstream reconfiguration support
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- EVRC decoder
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- EVRC decoder
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- audio fade filter
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version 1.1:
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version 1.1:
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@ -282,6 +282,83 @@ aconvert=u8:auto
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@end example
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@end example
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@end itemize
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@end itemize
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@section afade
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Apply fade-in/out effect to input audio.
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The filter accepts parameters as a list of @var{key}=@var{value}
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pairs, separated by ":".
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A description of the accepted parameters follows.
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@table @option
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@item type, t
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Specify the effect type, can be either @code{in} for fade-in, or
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@code{out} for a fade-out effect. Default is @code{in}.
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@item start_sample, ss
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Specify the number of the start sample for starting to apply the fade
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effect. Default is 0.
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@item nb_samples, ns
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Specify the number of samples for which the fade effect has to last. At
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the end of the fade-in effect the output audio will have the same
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volume as the input audio, at the end of the fade-out transition
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the output audio will be silence. Default is 44100.
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@item start_time, st
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Specify time in seconds for starting to apply the fade
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effect. Default is 0.
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If set this option is used instead of @var{start_sample} one.
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@item duration, d
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Specify the number of seconds for which the fade effect has to last. At
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the end of the fade-in effect the output audio will have the same
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volume as the input audio, at the end of the fade-out transition
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the output audio will be silence. Default is 0.
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If set this option is used instead of @var{nb_samples} one.
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@item curve
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Set cuve for fade transition.
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@table @option
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@item @var{triangular, linear slope (default)}
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@code{tri}
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@item @var{quarter of sine wave}
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@code{qsin}
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@item @var{half of sine wave}
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@code{esin}
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@item @var{exponential sine wave}
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@code{hsin}
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@item @var{logarithmic}
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@code{log}
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@item @var{inverted parabola}
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@code{par}
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@item @var{quadratic}
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@code{qua}
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@item @var{cubic}
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@code{cub}
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@item @var{square root}
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@code{squ}
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@item @var{cubic root}
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@code{cbr}
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@end table
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@end table
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@subsection Examples
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@itemize
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@item
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Fade in first 15 seconds of audio:
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@example
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afade=t=in:ss=0:d=15
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@end example
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@item
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Fade out last 25 seconds of a 900 seconds audio:
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@example
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afade=t=out:ss=875:d=25
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@end example
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@end itemize
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@section aformat
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@section aformat
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Set output format constraints for the input audio. The framework will
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Set output format constraints for the input audio. The framework will
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@ -51,6 +51,7 @@ OBJS-$(CONFIG_AVFORMAT) += lavfutils.o
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OBJS-$(CONFIG_SWSCALE) += lswsutils.o
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OBJS-$(CONFIG_SWSCALE) += lswsutils.o
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OBJS-$(CONFIG_ACONVERT_FILTER) += af_aconvert.o
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OBJS-$(CONFIG_ACONVERT_FILTER) += af_aconvert.o
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OBJS-$(CONFIG_AFADE_FILTER) += af_afade.o
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OBJS-$(CONFIG_AFORMAT_FILTER) += af_aformat.o
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OBJS-$(CONFIG_AFORMAT_FILTER) += af_aformat.o
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OBJS-$(CONFIG_AMERGE_FILTER) += af_amerge.o
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OBJS-$(CONFIG_AMERGE_FILTER) += af_amerge.o
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OBJS-$(CONFIG_AMIX_FILTER) += af_amix.o
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OBJS-$(CONFIG_AMIX_FILTER) += af_amix.o
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307
libavfilter/af_afade.c
Normal file
307
libavfilter/af_afade.c
Normal file
@ -0,0 +1,307 @@
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/*
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* Copyright (c) 2013 Paul B Mahol
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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/**
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* @file
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* fade audio filter
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*/
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#include "libavutil/opt.h"
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#include "audio.h"
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#include "avfilter.h"
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#include "internal.h"
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typedef struct {
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const AVClass *class;
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int type;
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int curve;
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int nb_samples;
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int64_t start_sample;
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double duration;
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double start_time;
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void (*fade_samples)(uint8_t **dst, uint8_t * const *src,
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int nb_samples, int channels, int direction,
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int64_t start, int range, int curve);
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} AudioFadeContext;
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enum CurveType { TRI, QSIN, ESIN, HSIN, LOG, PAR, QUA, CUB, SQU, CBR };
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#define OFFSET(x) offsetof(AudioFadeContext, x)
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#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
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static const AVOption afade_options[] = {
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{ "type", "set the fade direction", OFFSET(type), AV_OPT_TYPE_INT, {.i64 = 0 }, 0, 1, FLAGS, "type" },
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{ "t", "set the fade direction", OFFSET(type), AV_OPT_TYPE_INT, {.i64 = 0 }, 0, 1, FLAGS, "type" },
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{ "in", NULL, 0, AV_OPT_TYPE_CONST, {.i64 = 0 }, 0, 0, FLAGS, "type" },
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{ "out", NULL, 0, AV_OPT_TYPE_CONST, {.i64 = 1 }, 0, 0, FLAGS, "type" },
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{ "start_sample", "set expression of sample to start fading", OFFSET(start_sample), AV_OPT_TYPE_INT64, {.i64 = 0 }, 0, INT64_MAX, FLAGS },
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{ "ss", "set expression of sample to start fading", OFFSET(start_sample), AV_OPT_TYPE_INT64, {.i64 = 0 }, 0, INT64_MAX, FLAGS },
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{ "nb_samples", "set expression for fade duration in samples", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64 = 44100}, 1, INT32_MAX, FLAGS },
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{ "ns", "set expression for fade duration in samples", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64 = 44100}, 1, INT32_MAX, FLAGS },
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{ "start_time", "set expression of second to start fading", OFFSET(start_time), AV_OPT_TYPE_DOUBLE, {.dbl = 0. }, 0, 7*24*60*60,FLAGS },
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{ "st", "set expression of second to start fading", OFFSET(start_time), AV_OPT_TYPE_DOUBLE, {.dbl = 0. }, 0, 7*24*60*60,FLAGS },
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{ "duration", "set expression for fade duration in seconds", OFFSET(duration), AV_OPT_TYPE_DOUBLE, {.dbl = 0. }, 0, 24*60*60, FLAGS },
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{ "d", "set expression for fade duration in seconds", OFFSET(duration), AV_OPT_TYPE_DOUBLE, {.dbl = 0. }, 0, 24*60*60, FLAGS },
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{ "curve", "set expression for fade curve", OFFSET(curve), AV_OPT_TYPE_INT, {.i64 = TRI }, TRI, CBR, FLAGS, "curve" },
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{ "c", "set expression for fade curve", OFFSET(curve), AV_OPT_TYPE_INT, {.i64 = TRI }, TRI, CBR, FLAGS, "curve" },
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{ "tri", "linear slope", 0, AV_OPT_TYPE_CONST, {.i64 = TRI }, 0, 0, FLAGS, "curve" },
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{ "qsin", "quarter of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = QSIN }, 0, 0, FLAGS, "curve" },
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{ "esin", "exponential sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = ESIN }, 0, 0, FLAGS, "curve" },
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{ "hsin", "half of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = HSIN }, 0, 0, FLAGS, "curve" },
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{ "log", "logarithmic", 0, AV_OPT_TYPE_CONST, {.i64 = LOG }, 0, 0, FLAGS, "curve" },
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{ "par", "inverted parabola", 0, AV_OPT_TYPE_CONST, {.i64 = PAR }, 0, 0, FLAGS, "curve" },
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{ "qua", "quadratic", 0, AV_OPT_TYPE_CONST, {.i64 = QUA }, 0, 0, FLAGS, "curve" },
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{ "cub", "cubic", 0, AV_OPT_TYPE_CONST, {.i64 = CUB }, 0, 0, FLAGS, "curve" },
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{ "squ", "square root", 0, AV_OPT_TYPE_CONST, {.i64 = SQU }, 0, 0, FLAGS, "curve" },
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{ "cbr", "cubic root", 0, AV_OPT_TYPE_CONST, {.i64 = CBR }, 0, 0, FLAGS, "curve" },
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{NULL},
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};
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AVFILTER_DEFINE_CLASS(afade);
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static av_cold int init(AVFilterContext *ctx, const char *args)
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{
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AudioFadeContext *afade = ctx->priv;
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int ret;
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afade->class = &afade_class;
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av_opt_set_defaults(afade);
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if ((ret = av_set_options_string(afade, args, "=", ":")) < 0)
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return ret;
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if (INT64_MAX - afade->nb_samples < afade->start_sample)
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return AVERROR(EINVAL);
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return 0;
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}
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static int query_formats(AVFilterContext *ctx)
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{
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AVFilterFormats *formats;
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AVFilterChannelLayouts *layouts;
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static const enum AVSampleFormat sample_fmts[] = {
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AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16P,
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AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32P,
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AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP,
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AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBLP,
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AV_SAMPLE_FMT_NONE
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};
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layouts = ff_all_channel_layouts();
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if (!layouts)
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return AVERROR(ENOMEM);
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ff_set_common_channel_layouts(ctx, layouts);
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formats = ff_make_format_list(sample_fmts);
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if (!formats)
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return AVERROR(ENOMEM);
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ff_set_common_formats(ctx, formats);
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formats = ff_all_samplerates();
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if (!formats)
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return AVERROR(ENOMEM);
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ff_set_common_samplerates(ctx, formats);
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return 0;
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}
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static double fade_gain(int curve, int64_t index, int range)
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{
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double gain;
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gain = FFMAX(0.0, FFMIN(1.0, 1.0 * index / range));
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switch (curve) {
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case QSIN:
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gain = sin(gain * M_PI / 2.0);
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break;
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case ESIN:
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gain = 1.0 - cos(M_PI / 4.0 * (pow(2.0*gain - 1, 3) + 1));
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break;
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case HSIN:
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gain = (1.0 - cos(gain * M_PI)) / 2.0;
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break;
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case LOG:
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gain = pow(0.1, (1 - gain) * 5.0);
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break;
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case PAR:
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gain = (1 - (1 - gain) * (1 - gain));
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break;
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case QUA:
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gain *= gain;
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break;
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case CUB:
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gain = gain * gain * gain;
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break;
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case SQU:
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gain = sqrt(gain);
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break;
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case CBR:
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gain = cbrt(gain);
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break;
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}
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return gain;
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}
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#define FADE_PLANAR(name, type) \
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static void fade_samples_## name ##p(uint8_t **dst, uint8_t * const *src, \
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int nb_samples, int channels, int dir, \
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int64_t start, int range, int curve) \
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{ \
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int i, c; \
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\
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for (i = 0; i < nb_samples; i++) { \
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double gain = fade_gain(curve, start + i * dir, range); \
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for (c = 0; c < channels; c++) { \
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type *d = (type *)dst[c]; \
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const type *s = (type *)src[c]; \
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\
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d[i] = s[i] * gain; \
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} \
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} \
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}
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#define FADE(name, type) \
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static void fade_samples_## name (uint8_t **dst, uint8_t * const *src, \
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int nb_samples, int channels, int dir, \
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int64_t start, int range, int curve) \
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{ \
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type *d = (type *)dst[0]; \
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const type *s = (type *)src[0]; \
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int i, c, k = 0; \
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\
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for (i = 0; i < nb_samples; i++) { \
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double gain = fade_gain(curve, start + i * dir, range); \
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for (c = 0; c < channels; c++, k++) \
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d[k] = s[k] * gain; \
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} \
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}
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FADE_PLANAR(dbl, double)
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FADE_PLANAR(flt, float)
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FADE_PLANAR(s16, int16_t)
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FADE_PLANAR(s32, int32_t)
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FADE(dbl, double)
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FADE(flt, float)
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FADE(s16, int16_t)
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FADE(s32, int32_t)
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static int config_output(AVFilterLink *outlink)
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{
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AVFilterContext *ctx = outlink->src;
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AudioFadeContext *afade = ctx->priv;
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AVFilterLink *inlink = ctx->inputs[0];
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switch (inlink->format) {
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case AV_SAMPLE_FMT_DBL: afade->fade_samples = fade_samples_dbl; break;
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case AV_SAMPLE_FMT_DBLP: afade->fade_samples = fade_samples_dblp; break;
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case AV_SAMPLE_FMT_FLT: afade->fade_samples = fade_samples_flt; break;
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case AV_SAMPLE_FMT_FLTP: afade->fade_samples = fade_samples_fltp; break;
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case AV_SAMPLE_FMT_S16: afade->fade_samples = fade_samples_s16; break;
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case AV_SAMPLE_FMT_S16P: afade->fade_samples = fade_samples_s16p; break;
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case AV_SAMPLE_FMT_S32: afade->fade_samples = fade_samples_s32; break;
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case AV_SAMPLE_FMT_S32P: afade->fade_samples = fade_samples_s32p; break;
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}
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if (afade->duration)
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afade->nb_samples = afade->duration * inlink->sample_rate;
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if (afade->start_time)
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afade->start_sample = afade->start_time * inlink->sample_rate;
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return 0;
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}
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||||||
|
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;
|
initialized = 1;
|
||||||
|
|
||||||
REGISTER_FILTER(ACONVERT, aconvert, af);
|
REGISTER_FILTER(ACONVERT, aconvert, af);
|
||||||
|
REGISTER_FILTER(AFADE, afade, af);
|
||||||
REGISTER_FILTER(AFORMAT, aformat, af);
|
REGISTER_FILTER(AFORMAT, aformat, af);
|
||||||
REGISTER_FILTER(AMERGE, amerge, af);
|
REGISTER_FILTER(AMERGE, amerge, af);
|
||||||
REGISTER_FILTER(AMIX, amix, af);
|
REGISTER_FILTER(AMIX, amix, af);
|
||||||
|
|||||||
@ -29,8 +29,8 @@
|
|||||||
#include "libavutil/avutil.h"
|
#include "libavutil/avutil.h"
|
||||||
|
|
||||||
#define LIBAVFILTER_VERSION_MAJOR 3
|
#define LIBAVFILTER_VERSION_MAJOR 3
|
||||||
#define LIBAVFILTER_VERSION_MINOR 32
|
#define LIBAVFILTER_VERSION_MINOR 33
|
||||||
#define LIBAVFILTER_VERSION_MICRO 101
|
#define LIBAVFILTER_VERSION_MICRO 100
|
||||||
|
|
||||||
#define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \
|
#define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \
|
||||||
LIBAVFILTER_VERSION_MINOR, \
|
LIBAVFILTER_VERSION_MINOR, \
|
||||||
|
|||||||
Loading…
Reference in New Issue
Block a user