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FFmpeg/libavfilter/af_afade.c
Niklas Haas dfc3256da1 avfilter/af_afade: support multiple inputs 
Instead of just 2 files, generalize this filter to support crossfading
arbitrarily many files. This makes the filter essentially operate similar
to the `concat` filter, chaining multiple files one after another.

Aside from just adding more input pads, this requires rewriting the
activate function. Instead of a finite state machine, we keep track of the
currently active input index; and advance it only once the current input is
fully exhausted.

This results in arguably simpler logic overall.
2025-09-04 23:12:34 +00:00

826 lines
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/*
* Copyright (c) 2013-2015 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 "config_components.h"
#include "libavutil/avassert.h"
#include "libavutil/avstring.h"
#include "libavutil/opt.h"
#include "audio.h"
#include "avfilter.h"
#include "filters.h"
typedef struct AudioFadeContext {
const AVClass *class;
int nb_inputs;
int type;
int curve, curve2;
int64_t nb_samples;
int64_t start_sample;
int64_t duration;
int64_t start_time;
double silence;
double unity;
int overlap;
int64_t pts;
int xfade_idx;
void (*fade_samples)(uint8_t **dst, uint8_t * const *src,
int nb_samples, int channels, int direction,
int64_t start, int64_t range, int curve,
double silence, double unity);
void (*scale_samples)(uint8_t **dst, uint8_t * const *src,
int nb_samples, int channels, double unity);
void (*crossfade_samples)(uint8_t **dst, uint8_t * const *cf0,
uint8_t * const *cf1,
int nb_samples, int channels,
int curve0, int curve1);
} AudioFadeContext;
enum CurveType { NONE = -1, TRI, QSIN, ESIN, HSIN, LOG, IPAR, QUA, CUB, SQU, CBR, PAR, EXP, IQSIN, IHSIN, DESE, DESI, LOSI, SINC, ISINC, QUAT, QUATR, QSIN2, HSIN2, NB_CURVES };
#define OFFSET(x) offsetof(AudioFadeContext, x)
#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
#define TFLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
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
};
static double fade_gain(int curve, int64_t index, int64_t range, double silence, double unity)
{
#define CUBE(a) ((a)*(a)*(a))
double gain;
gain = av_clipd(1.0 * index / range, 0, 1.0);
switch (curve) {
case QSIN:
gain = sin(gain * M_PI / 2.0);
break;
case IQSIN:
/* 0.6... = 2 / M_PI */
gain = 0.6366197723675814 * asin(gain);
break;
case ESIN:
gain = 1.0 - cos(M_PI / 4.0 * (CUBE(2.0*gain - 1) + 1));
break;
case HSIN:
gain = (1.0 - cos(gain * M_PI)) / 2.0;
break;
case IHSIN:
/* 0.3... = 1 / M_PI */
gain = 0.3183098861837907 * acos(1 - 2 * gain);
break;
case EXP:
/* -11.5... = 5*ln(0.1) */
gain = exp(-11.512925464970227 * (1 - gain));
break;
case LOG:
gain = av_clipd(1 + 0.2 * log10(gain), 0, 1.0);
break;
case PAR:
gain = 1 - sqrt(1 - gain);
break;
case IPAR:
gain = (1 - (1 - gain) * (1 - gain));
break;
case QUA:
gain *= gain;
break;
case CUB:
gain = CUBE(gain);
break;
case SQU:
gain = sqrt(gain);
break;
case CBR:
gain = cbrt(gain);
break;
case DESE:
gain = gain <= 0.5 ? cbrt(2 * gain) / 2: 1 - cbrt(2 * (1 - gain)) / 2;
break;
case DESI:
gain = gain <= 0.5 ? CUBE(2 * gain) / 2: 1 - CUBE(2 * (1 - gain)) / 2;
break;
case LOSI: {
const double a = 1. / (1. - 0.787) - 1;
double A = 1. / (1.0 + exp(0 -((gain-0.5) * a * 2.0)));
double B = 1. / (1.0 + exp(a));
double C = 1. / (1.0 + exp(0-a));
gain = (A - B) / (C - B);
}
break;
case SINC:
gain = gain >= 1.0 ? 1.0 : sin(M_PI * (1.0 - gain)) / (M_PI * (1.0 - gain));
break;
case ISINC:
gain = gain <= 0.0 ? 0.0 : 1.0 - sin(M_PI * gain) / (M_PI * gain);
break;
case QUAT:
gain = gain * gain * gain * gain;
break;
case QUATR:
gain = pow(gain, 0.25);
break;
case QSIN2:
gain = sin(gain * M_PI / 2.0) * sin(gain * M_PI / 2.0);
break;
case HSIN2:
gain = pow((1.0 - cos(gain * M_PI)) / 2.0, 2.0);
break;
case NONE:
gain = 1.0;
break;
}
return silence + (unity - silence) * 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, int64_t range,int curve,\
double silence, double unity) \
{ \
int i, c; \
\
for (i = 0; i < nb_samples; i++) { \
double gain = fade_gain(curve, start + i * dir,range,silence,unity);\
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, int64_t range, int curve, \
double silence, double unity) \
{ \
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,silence,unity);\
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)
#define SCALE_PLANAR(name, type) \
static void scale_samples_## name ##p(uint8_t **dst, uint8_t * const *src, \
int nb_samples, int channels, \
double gain) \
{ \
int i, c; \
\
for (i = 0; i < nb_samples; i++) { \
for (c = 0; c < channels; c++) { \
type *d = (type *)dst[c]; \
const type *s = (type *)src[c]; \
\
d[i] = s[i] * gain; \
} \
} \
}
#define SCALE(name, type) \
static void scale_samples_## name (uint8_t **dst, uint8_t * const *src, \
int nb_samples, int channels, double gain)\
{ \
type *d = (type *)dst[0]; \
const type *s = (type *)src[0]; \
int i, c, k = 0; \
\
for (i = 0; i < nb_samples; i++) { \
for (c = 0; c < channels; c++, k++) \
d[k] = s[k] * gain; \
} \
}
SCALE_PLANAR(dbl, double)
SCALE_PLANAR(flt, float)
SCALE_PLANAR(s16, int16_t)
SCALE_PLANAR(s32, int32_t)
SCALE(dbl, double)
SCALE(flt, float)
SCALE(s16, int16_t)
SCALE(s32, int32_t)
static int config_output(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
AudioFadeContext *s = ctx->priv;
switch (outlink->format) {
case AV_SAMPLE_FMT_DBL: s->fade_samples = fade_samples_dbl;
s->scale_samples = scale_samples_dbl;
break;
case AV_SAMPLE_FMT_DBLP: s->fade_samples = fade_samples_dblp;
s->scale_samples = scale_samples_dblp;
break;
case AV_SAMPLE_FMT_FLT: s->fade_samples = fade_samples_flt;
s->scale_samples = scale_samples_flt;
break;
case AV_SAMPLE_FMT_FLTP: s->fade_samples = fade_samples_fltp;
s->scale_samples = scale_samples_fltp;
break;
case AV_SAMPLE_FMT_S16: s->fade_samples = fade_samples_s16;
s->scale_samples = scale_samples_s16;
break;
case AV_SAMPLE_FMT_S16P: s->fade_samples = fade_samples_s16p;
s->scale_samples = scale_samples_s16p;
break;
case AV_SAMPLE_FMT_S32: s->fade_samples = fade_samples_s32;
s->scale_samples = scale_samples_s32;
break;
case AV_SAMPLE_FMT_S32P: s->fade_samples = fade_samples_s32p;
s->scale_samples = scale_samples_s32p;
break;
}
if (s->duration)
s->nb_samples = av_rescale(s->duration, outlink->sample_rate, AV_TIME_BASE);
s->duration = 0;
if (s->start_time)
s->start_sample = av_rescale(s->start_time, outlink->sample_rate, AV_TIME_BASE);
s->start_time = 0;
return 0;
}
#if CONFIG_AFADE_FILTER
static const AVOption afade_options[] = {
{ "type", "set the fade direction", OFFSET(type), AV_OPT_TYPE_INT, {.i64 = 0 }, 0, 1, TFLAGS, .unit = "type" },
{ "t", "set the fade direction", OFFSET(type), AV_OPT_TYPE_INT, {.i64 = 0 }, 0, 1, TFLAGS, .unit = "type" },
{ "in", "fade-in", 0, AV_OPT_TYPE_CONST, {.i64 = 0 }, 0, 0, TFLAGS, .unit = "type" },
{ "out", "fade-out", 0, AV_OPT_TYPE_CONST, {.i64 = 1 }, 0, 0, TFLAGS, .unit = "type" },
{ "start_sample", "set number of first sample to start fading", OFFSET(start_sample), AV_OPT_TYPE_INT64, {.i64 = 0 }, 0, INT64_MAX, TFLAGS },
{ "ss", "set number of first sample to start fading", OFFSET(start_sample), AV_OPT_TYPE_INT64, {.i64 = 0 }, 0, INT64_MAX, TFLAGS },
{ "nb_samples", "set number of samples for fade duration", OFFSET(nb_samples), AV_OPT_TYPE_INT64, {.i64 = 44100}, 1, INT64_MAX, TFLAGS },
{ "ns", "set number of samples for fade duration", OFFSET(nb_samples), AV_OPT_TYPE_INT64, {.i64 = 44100}, 1, INT64_MAX, TFLAGS },
{ "start_time", "set time to start fading", OFFSET(start_time), AV_OPT_TYPE_DURATION, {.i64 = 0 }, 0, INT64_MAX, TFLAGS },
{ "st", "set time to start fading", OFFSET(start_time), AV_OPT_TYPE_DURATION, {.i64 = 0 }, 0, INT64_MAX, TFLAGS },
{ "duration", "set fade duration", OFFSET(duration), AV_OPT_TYPE_DURATION, {.i64 = 0 }, 0, INT64_MAX, TFLAGS },
{ "d", "set fade duration", OFFSET(duration), AV_OPT_TYPE_DURATION, {.i64 = 0 }, 0, INT64_MAX, TFLAGS },
{ "curve", "set fade curve type", OFFSET(curve), AV_OPT_TYPE_INT, {.i64 = TRI }, NONE, NB_CURVES - 1, TFLAGS, .unit = "curve" },
{ "c", "set fade curve type", OFFSET(curve), AV_OPT_TYPE_INT, {.i64 = TRI }, NONE, NB_CURVES - 1, TFLAGS, .unit = "curve" },
{ "nofade", "no fade; keep audio as-is", 0, AV_OPT_TYPE_CONST, {.i64 = NONE }, 0, 0, TFLAGS, .unit = "curve" },
{ "tri", "linear slope", 0, AV_OPT_TYPE_CONST, {.i64 = TRI }, 0, 0, TFLAGS, .unit = "curve" },
{ "qsin", "quarter of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = QSIN }, 0, 0, TFLAGS, .unit = "curve" },
{ "esin", "exponential sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = ESIN }, 0, 0, TFLAGS, .unit = "curve" },
{ "hsin", "half of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = HSIN }, 0, 0, TFLAGS, .unit = "curve" },
{ "log", "logarithmic", 0, AV_OPT_TYPE_CONST, {.i64 = LOG }, 0, 0, TFLAGS, .unit = "curve" },
{ "ipar", "inverted parabola", 0, AV_OPT_TYPE_CONST, {.i64 = IPAR }, 0, 0, TFLAGS, .unit = "curve" },
{ "qua", "quadratic", 0, AV_OPT_TYPE_CONST, {.i64 = QUA }, 0, 0, TFLAGS, .unit = "curve" },
{ "cub", "cubic", 0, AV_OPT_TYPE_CONST, {.i64 = CUB }, 0, 0, TFLAGS, .unit = "curve" },
{ "squ", "square root", 0, AV_OPT_TYPE_CONST, {.i64 = SQU }, 0, 0, TFLAGS, .unit = "curve" },
{ "cbr", "cubic root", 0, AV_OPT_TYPE_CONST, {.i64 = CBR }, 0, 0, TFLAGS, .unit = "curve" },
{ "par", "parabola", 0, AV_OPT_TYPE_CONST, {.i64 = PAR }, 0, 0, TFLAGS, .unit = "curve" },
{ "exp", "exponential", 0, AV_OPT_TYPE_CONST, {.i64 = EXP }, 0, 0, TFLAGS, .unit = "curve" },
{ "iqsin", "inverted quarter of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = IQSIN}, 0, 0, TFLAGS, .unit = "curve" },
{ "ihsin", "inverted half of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = IHSIN}, 0, 0, TFLAGS, .unit = "curve" },
{ "dese", "double-exponential seat", 0, AV_OPT_TYPE_CONST, {.i64 = DESE }, 0, 0, TFLAGS, .unit = "curve" },
{ "desi", "double-exponential sigmoid", 0, AV_OPT_TYPE_CONST, {.i64 = DESI }, 0, 0, TFLAGS, .unit = "curve" },
{ "losi", "logistic sigmoid", 0, AV_OPT_TYPE_CONST, {.i64 = LOSI }, 0, 0, TFLAGS, .unit = "curve" },
{ "sinc", "sine cardinal function", 0, AV_OPT_TYPE_CONST, {.i64 = SINC }, 0, 0, TFLAGS, .unit = "curve" },
{ "isinc", "inverted sine cardinal function", 0, AV_OPT_TYPE_CONST, {.i64 = ISINC}, 0, 0, TFLAGS, .unit = "curve" },
{ "quat", "quartic", 0, AV_OPT_TYPE_CONST, {.i64 = QUAT }, 0, 0, TFLAGS, .unit = "curve" },
{ "quatr", "quartic root", 0, AV_OPT_TYPE_CONST, {.i64 = QUATR}, 0, 0, TFLAGS, .unit = "curve" },
{ "qsin2", "squared quarter of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = QSIN2}, 0, 0, TFLAGS, .unit = "curve" },
{ "hsin2", "squared half of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = HSIN2}, 0, 0, TFLAGS, .unit = "curve" },
{ "silence", "set the silence gain", OFFSET(silence), AV_OPT_TYPE_DOUBLE, {.dbl = 0 }, 0, 1, TFLAGS },
{ "unity", "set the unity gain", OFFSET(unity), AV_OPT_TYPE_DOUBLE, {.dbl = 1 }, 0, 1, TFLAGS },
{ NULL }
};
AVFILTER_DEFINE_CLASS(afade);
static av_cold int init(AVFilterContext *ctx)
{
AudioFadeContext *s = ctx->priv;
if (INT64_MAX - s->nb_samples < s->start_sample)
return AVERROR(EINVAL);
return 0;
}
static int filter_frame(AVFilterLink *inlink, AVFrame *buf)
{
AudioFadeContext *s = inlink->dst->priv;
AVFilterLink *outlink = inlink->dst->outputs[0];
int nb_samples = buf->nb_samples;
AVFrame *out_buf;
int64_t cur_sample = av_rescale_q(buf->pts, inlink->time_base, (AVRational){1, inlink->sample_rate});
if (s->unity == 1.0 &&
((!s->type && (s->start_sample + s->nb_samples < cur_sample)) ||
( s->type && (cur_sample + nb_samples < s->start_sample))))
return ff_filter_frame(outlink, buf);
if (av_frame_is_writable(buf)) {
out_buf = buf;
} else {
out_buf = ff_get_audio_buffer(outlink, nb_samples);
if (!out_buf)
return AVERROR(ENOMEM);
av_frame_copy_props(out_buf, buf);
}
if ((!s->type && (cur_sample + nb_samples < s->start_sample)) ||
( s->type && (s->start_sample + s->nb_samples < cur_sample))) {
if (s->silence == 0.) {
av_samples_set_silence(out_buf->extended_data, 0, nb_samples,
out_buf->ch_layout.nb_channels, out_buf->format);
} else {
s->scale_samples(out_buf->extended_data, buf->extended_data,
nb_samples, buf->ch_layout.nb_channels,
s->silence);
}
} else if (( s->type && (cur_sample + nb_samples < s->start_sample)) ||
(!s->type && (s->start_sample + s->nb_samples < cur_sample))) {
s->scale_samples(out_buf->extended_data, buf->extended_data,
nb_samples, buf->ch_layout.nb_channels,
s->unity);
} else {
int64_t start;
if (!s->type)
start = cur_sample - s->start_sample;
else
start = s->start_sample + s->nb_samples - cur_sample;
s->fade_samples(out_buf->extended_data, buf->extended_data,
nb_samples, buf->ch_layout.nb_channels,
s->type ? -1 : 1, start,
s->nb_samples, s->curve, s->silence, s->unity);
}
if (buf != out_buf)
av_frame_free(&buf);
return ff_filter_frame(outlink, out_buf);
}
static int process_command(AVFilterContext *ctx, const char *cmd, const char *args,
char *res, int res_len, int flags)
{
int ret;
ret = ff_filter_process_command(ctx, cmd, args, res, res_len, flags);
if (ret < 0)
return ret;
return config_output(ctx->outputs[0]);
}
static const AVFilterPad avfilter_af_afade_inputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.filter_frame = filter_frame,
},
};
static const AVFilterPad avfilter_af_afade_outputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_output,
},
};
const FFFilter ff_af_afade = {
.p.name = "afade",
.p.description = NULL_IF_CONFIG_SMALL("Fade in/out input audio."),
.p.priv_class = &afade_class,
.p.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC,
.priv_size = sizeof(AudioFadeContext),
.init = init,
FILTER_INPUTS(avfilter_af_afade_inputs),
FILTER_OUTPUTS(avfilter_af_afade_outputs),
FILTER_SAMPLEFMTS_ARRAY(sample_fmts),
.process_command = process_command,
};
#endif /* CONFIG_AFADE_FILTER */
#if CONFIG_ACROSSFADE_FILTER
static const AVOption acrossfade_options[] = {
{ "inputs", "set number of input files to cross fade", OFFSET(nb_inputs), AV_OPT_TYPE_INT, {.i64 = 2}, 1, INT32_MAX, FLAGS },
{ "n", "set number of input files to cross fade", OFFSET(nb_inputs), AV_OPT_TYPE_INT, {.i64 = 2}, 1, INT32_MAX, FLAGS },
{ "nb_samples", "set number of samples for cross fade duration", OFFSET(nb_samples), AV_OPT_TYPE_INT64, {.i64 = 44100}, 1, INT32_MAX/10, FLAGS },
{ "ns", "set number of samples for cross fade duration", OFFSET(nb_samples), AV_OPT_TYPE_INT64, {.i64 = 44100}, 1, INT32_MAX/10, FLAGS },
{ "duration", "set cross fade duration", OFFSET(duration), AV_OPT_TYPE_DURATION, {.i64 = 0 }, 0, 60000000, FLAGS },
{ "d", "set cross fade duration", OFFSET(duration), AV_OPT_TYPE_DURATION, {.i64 = 0 }, 0, 60000000, FLAGS },
{ "overlap", "overlap 1st stream end with 2nd stream start", OFFSET(overlap), AV_OPT_TYPE_BOOL, {.i64 = 1 }, 0, 1, FLAGS },
{ "o", "overlap 1st stream end with 2nd stream start", OFFSET(overlap), AV_OPT_TYPE_BOOL, {.i64 = 1 }, 0, 1, FLAGS },
{ "curve1", "set fade curve type for 1st stream", OFFSET(curve), AV_OPT_TYPE_INT, {.i64 = TRI }, NONE, NB_CURVES - 1, FLAGS, .unit = "curve" },
{ "c1", "set fade curve type for 1st stream", OFFSET(curve), AV_OPT_TYPE_INT, {.i64 = TRI }, NONE, NB_CURVES - 1, FLAGS, .unit = "curve" },
{ "nofade", "no fade; keep audio as-is", 0, AV_OPT_TYPE_CONST, {.i64 = NONE }, 0, 0, FLAGS, .unit = "curve" },
{ "tri", "linear slope", 0, AV_OPT_TYPE_CONST, {.i64 = TRI }, 0, 0, FLAGS, .unit = "curve" },
{ "qsin", "quarter of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = QSIN }, 0, 0, FLAGS, .unit = "curve" },
{ "esin", "exponential sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = ESIN }, 0, 0, FLAGS, .unit = "curve" },
{ "hsin", "half of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = HSIN }, 0, 0, FLAGS, .unit = "curve" },
{ "log", "logarithmic", 0, AV_OPT_TYPE_CONST, {.i64 = LOG }, 0, 0, FLAGS, .unit = "curve" },
{ "ipar", "inverted parabola", 0, AV_OPT_TYPE_CONST, {.i64 = IPAR }, 0, 0, FLAGS, .unit = "curve" },
{ "qua", "quadratic", 0, AV_OPT_TYPE_CONST, {.i64 = QUA }, 0, 0, FLAGS, .unit = "curve" },
{ "cub", "cubic", 0, AV_OPT_TYPE_CONST, {.i64 = CUB }, 0, 0, FLAGS, .unit = "curve" },
{ "squ", "square root", 0, AV_OPT_TYPE_CONST, {.i64 = SQU }, 0, 0, FLAGS, .unit = "curve" },
{ "cbr", "cubic root", 0, AV_OPT_TYPE_CONST, {.i64 = CBR }, 0, 0, FLAGS, .unit = "curve" },
{ "par", "parabola", 0, AV_OPT_TYPE_CONST, {.i64 = PAR }, 0, 0, FLAGS, .unit = "curve" },
{ "exp", "exponential", 0, AV_OPT_TYPE_CONST, {.i64 = EXP }, 0, 0, FLAGS, .unit = "curve" },
{ "iqsin", "inverted quarter of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = IQSIN}, 0, 0, FLAGS, .unit = "curve" },
{ "ihsin", "inverted half of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = IHSIN}, 0, 0, FLAGS, .unit = "curve" },
{ "dese", "double-exponential seat", 0, AV_OPT_TYPE_CONST, {.i64 = DESE }, 0, 0, FLAGS, .unit = "curve" },
{ "desi", "double-exponential sigmoid", 0, AV_OPT_TYPE_CONST, {.i64 = DESI }, 0, 0, FLAGS, .unit = "curve" },
{ "losi", "logistic sigmoid", 0, AV_OPT_TYPE_CONST, {.i64 = LOSI }, 0, 0, FLAGS, .unit = "curve" },
{ "sinc", "sine cardinal function", 0, AV_OPT_TYPE_CONST, {.i64 = SINC }, 0, 0, FLAGS, .unit = "curve" },
{ "isinc", "inverted sine cardinal function", 0, AV_OPT_TYPE_CONST, {.i64 = ISINC}, 0, 0, FLAGS, .unit = "curve" },
{ "quat", "quartic", 0, AV_OPT_TYPE_CONST, {.i64 = QUAT }, 0, 0, FLAGS, .unit = "curve" },
{ "quatr", "quartic root", 0, AV_OPT_TYPE_CONST, {.i64 = QUATR}, 0, 0, FLAGS, .unit = "curve" },
{ "qsin2", "squared quarter of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = QSIN2}, 0, 0, FLAGS, .unit = "curve" },
{ "hsin2", "squared half of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = HSIN2}, 0, 0, FLAGS, .unit = "curve" },
{ "curve2", "set fade curve type for 2nd stream", OFFSET(curve2), AV_OPT_TYPE_INT, {.i64 = TRI }, NONE, NB_CURVES - 1, FLAGS, .unit = "curve" },
{ "c2", "set fade curve type for 2nd stream", OFFSET(curve2), AV_OPT_TYPE_INT, {.i64 = TRI }, NONE, NB_CURVES - 1, FLAGS, .unit = "curve" },
{ NULL }
};
AVFILTER_DEFINE_CLASS(acrossfade);
#define CROSSFADE_PLANAR(name, type) \
static void crossfade_samples_## name ##p(uint8_t **dst, uint8_t * const *cf0, \
uint8_t * const *cf1, \
int nb_samples, int channels, \
int curve0, int curve1) \
{ \
int i, c; \
\
for (i = 0; i < nb_samples; i++) { \
double gain0 = fade_gain(curve0, nb_samples - 1 - i, nb_samples,0.,1.);\
double gain1 = fade_gain(curve1, i, nb_samples, 0., 1.); \
for (c = 0; c < channels; c++) { \
type *d = (type *)dst[c]; \
const type *s0 = (type *)cf0[c]; \
const type *s1 = (type *)cf1[c]; \
\
d[i] = s0[i] * gain0 + s1[i] * gain1; \
} \
} \
}
#define CROSSFADE(name, type) \
static void crossfade_samples_## name (uint8_t **dst, uint8_t * const *cf0, \
uint8_t * const *cf1, \
int nb_samples, int channels, \
int curve0, int curve1) \
{ \
type *d = (type *)dst[0]; \
const type *s0 = (type *)cf0[0]; \
const type *s1 = (type *)cf1[0]; \
int i, c, k = 0; \
\
for (i = 0; i < nb_samples; i++) { \
double gain0 = fade_gain(curve0, nb_samples - 1-i,nb_samples,0.,1.);\
double gain1 = fade_gain(curve1, i, nb_samples, 0., 1.); \
for (c = 0; c < channels; c++, k++) \
d[k] = s0[k] * gain0 + s1[k] * gain1; \
} \
}
CROSSFADE_PLANAR(dbl, double)
CROSSFADE_PLANAR(flt, float)
CROSSFADE_PLANAR(s16, int16_t)
CROSSFADE_PLANAR(s32, int32_t)
CROSSFADE(dbl, double)
CROSSFADE(flt, float)
CROSSFADE(s16, int16_t)
CROSSFADE(s32, int32_t)
static int pass_frame(AVFilterLink *inlink, AVFilterLink *outlink, int64_t *pts)
{
AVFrame *in;
int ret = ff_inlink_consume_frame(inlink, &in);
if (ret < 0)
return ret;
av_assert1(ret);
in->pts = *pts;
*pts += av_rescale_q(in->nb_samples,
(AVRational){ 1, outlink->sample_rate }, outlink->time_base);
return ff_filter_frame(outlink, in);
}
static int pass_samples(AVFilterLink *inlink, AVFilterLink *outlink, unsigned nb_samples, int64_t *pts)
{
AVFrame *in;
int ret = ff_inlink_consume_samples(inlink, nb_samples, nb_samples, &in);
if (ret < 0)
return ret;
av_assert1(ret);
in->pts = *pts;
*pts += av_rescale_q(in->nb_samples,
(AVRational){ 1, outlink->sample_rate }, outlink->time_base);
return ff_filter_frame(outlink, in);
}
static int pass_crossfade(AVFilterContext *ctx, const int idx0, const int idx1)
{
AudioFadeContext *s = ctx->priv;
AVFilterLink *outlink = ctx->outputs[0];
AVFrame *out, *cf[2] = { NULL };
int ret;
AVFilterLink *in0 = ctx->inputs[idx0];
AVFilterLink *in1 = ctx->inputs[idx1];
int queued_samples0 = ff_inlink_queued_samples(in0);
int queued_samples1 = ff_inlink_queued_samples(in1);
/* Limit to the relevant region */
av_assert1(queued_samples0 <= s->nb_samples);
if (ff_outlink_get_status(in1) && idx1 < s->nb_inputs - 1)
queued_samples1 /= 2; /* reserve second half for next fade-out */
queued_samples1 = FFMIN(queued_samples1, s->nb_samples);
if (s->overlap) {
int nb_samples = FFMIN(queued_samples0, queued_samples1);
if (nb_samples < s->nb_samples) {
av_log(ctx, AV_LOG_WARNING, "Input %d duration (%d samples) "
"is shorter than crossfade duration (%"PRId64" samples), "
"crossfade will be shorter by %"PRId64" samples.\n",
queued_samples0 <= queued_samples1 ? idx0 : idx1,
nb_samples, s->nb_samples, s->nb_samples - nb_samples);
if (queued_samples0 > nb_samples) {
ret = pass_samples(in0, outlink, queued_samples0 - nb_samples, &s->pts);
if (ret < 0)
return ret;
}
if (!nb_samples)
return 0; /* either input was completely empty */
}
av_assert1(nb_samples > 0);
out = ff_get_audio_buffer(outlink, nb_samples);
if (!out)
return AVERROR(ENOMEM);
ret = ff_inlink_consume_samples(in0, nb_samples, nb_samples, &cf[0]);
if (ret < 0) {
av_frame_free(&out);
return ret;
}
ret = ff_inlink_consume_samples(in1, nb_samples, nb_samples, &cf[1]);
if (ret < 0) {
av_frame_free(&cf[0]);
av_frame_free(&out);
return ret;
}
s->crossfade_samples(out->extended_data, cf[0]->extended_data,
cf[1]->extended_data, nb_samples,
out->ch_layout.nb_channels, s->curve, s->curve2);
out->pts = s->pts;
s->pts += av_rescale_q(nb_samples,
(AVRational){ 1, outlink->sample_rate }, outlink->time_base);
av_frame_free(&cf[0]);
av_frame_free(&cf[1]);
return ff_filter_frame(outlink, out);
} else {
if (queued_samples0 < s->nb_samples) {
av_log(ctx, AV_LOG_WARNING, "Input %d duration (%d samples) "
"is shorter than crossfade duration (%"PRId64" samples), "
"fade-out will be shorter by %"PRId64" samples.\n",
idx0, queued_samples0, s->nb_samples,
s->nb_samples - queued_samples0);
if (!queued_samples0)
goto fade_in;
}
out = ff_get_audio_buffer(outlink, queued_samples0);
if (!out)
return AVERROR(ENOMEM);
ret = ff_inlink_consume_samples(in0, queued_samples0, queued_samples0, &cf[0]);
if (ret < 0) {
av_frame_free(&out);
return ret;
}
s->fade_samples(out->extended_data, cf[0]->extended_data, cf[0]->nb_samples,
outlink->ch_layout.nb_channels, -1, cf[0]->nb_samples - 1, cf[0]->nb_samples, s->curve, 0., 1.);
out->pts = s->pts;
s->pts += av_rescale_q(cf[0]->nb_samples,
(AVRational){ 1, outlink->sample_rate }, outlink->time_base);
av_frame_free(&cf[0]);
ret = ff_filter_frame(outlink, out);
if (ret < 0)
return ret;
fade_in:
if (queued_samples1 < s->nb_samples) {
av_log(ctx, AV_LOG_WARNING, "Input %d duration (%d samples) "
"is shorter than crossfade duration (%"PRId64" samples), "
"fade-in will be shorter by %"PRId64" samples.\n",
idx1, ff_inlink_queued_samples(in1), s->nb_samples,
s->nb_samples - queued_samples1);
if (!queued_samples1)
return 0;
}
out = ff_get_audio_buffer(outlink, queued_samples1);
if (!out)
return AVERROR(ENOMEM);
ret = ff_inlink_consume_samples(in1, queued_samples1, queued_samples1, &cf[1]);
if (ret < 0) {
av_frame_free(&out);
return ret;
}
s->fade_samples(out->extended_data, cf[1]->extended_data, cf[1]->nb_samples,
outlink->ch_layout.nb_channels, 1, 0, cf[1]->nb_samples, s->curve2, 0., 1.);
out->pts = s->pts;
s->pts += av_rescale_q(cf[1]->nb_samples,
(AVRational){ 1, outlink->sample_rate }, outlink->time_base);
av_frame_free(&cf[1]);
return ff_filter_frame(outlink, out);
}
}
static int activate(AVFilterContext *ctx)
{
AudioFadeContext *s = ctx->priv;
const int idx0 = s->xfade_idx;
const int idx1 = s->xfade_idx + 1;
AVFilterLink *outlink = ctx->outputs[0];
AVFilterLink *in0 = ctx->inputs[idx0];
FF_FILTER_FORWARD_STATUS_BACK_ALL(outlink, ctx);
if (idx0 == s->nb_inputs - 1) {
/* Last active input, read until EOF */
if (ff_inlink_queued_frames(in0))
return pass_frame(in0, outlink, &s->pts);
FF_FILTER_FORWARD_STATUS(in0, outlink);
FF_FILTER_FORWARD_WANTED(outlink, in0);
return FFERROR_NOT_READY;
}
AVFilterLink *in1 = ctx->inputs[idx1];
int queued_samples0 = ff_inlink_queued_samples(in0);
if (queued_samples0 > s->nb_samples) {
AVFrame *frame = ff_inlink_peek_frame(in0, 0);
if (queued_samples0 - s->nb_samples >= frame->nb_samples)
return pass_frame(in0, outlink, &s->pts);
}
/* Continue reading until EOF */
if (ff_outlink_get_status(in0)) {
if (queued_samples0 > s->nb_samples)
return pass_samples(in0, outlink, queued_samples0 - s->nb_samples, &s->pts);
} else {
FF_FILTER_FORWARD_WANTED(outlink, in0);
return FFERROR_NOT_READY;
}
/* At this point, in0 has reached EOF with no more samples remaining
* except those that we want to crossfade */
av_assert0(queued_samples0 <= s->nb_samples);
int queued_samples1 = ff_inlink_queued_samples(in1);
/* If this clip is sandwiched between two other clips, buffer at least
* twice the total crossfade duration to ensure that we won't reach EOF
* during the second fade (in which case we would shorten the fade) */
int needed_samples = s->nb_samples;
if (idx1 < s->nb_inputs - 1)
needed_samples *= 2;
if (queued_samples1 >= needed_samples || ff_outlink_get_status(in1)) {
/* The first filter may EOF before delivering any samples, in which
* case it's possible for pass_crossfade() to be a no-op. Just ensure
* the activate() function runs again after incrementing the index to
* ensure we correctly move on to the next input in that case. */
s->xfade_idx++;
ff_filter_set_ready(ctx, 10);
return pass_crossfade(ctx, idx0, idx1);
} else {
FF_FILTER_FORWARD_WANTED(outlink, in1);
return FFERROR_NOT_READY;
}
}
static av_cold int acrossfade_init(AVFilterContext *ctx)
{
AudioFadeContext *s = ctx->priv;
int ret;
for (int i = 0; i < s->nb_inputs; i++) {
AVFilterPad pad = {
.name = av_asprintf("crossfade%d", i),
.type = AVMEDIA_TYPE_AUDIO,
};
if (!pad.name)
return AVERROR(ENOMEM);
ret = ff_append_inpad_free_name(ctx, &pad);
if (ret < 0)
return ret;
}
return 0;
}
static int acrossfade_config_output(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
AudioFadeContext *s = ctx->priv;
outlink->time_base = ctx->inputs[0]->time_base;
switch (outlink->format) {
case AV_SAMPLE_FMT_DBL: s->crossfade_samples = crossfade_samples_dbl; break;
case AV_SAMPLE_FMT_DBLP: s->crossfade_samples = crossfade_samples_dblp; break;
case AV_SAMPLE_FMT_FLT: s->crossfade_samples = crossfade_samples_flt; break;
case AV_SAMPLE_FMT_FLTP: s->crossfade_samples = crossfade_samples_fltp; break;
case AV_SAMPLE_FMT_S16: s->crossfade_samples = crossfade_samples_s16; break;
case AV_SAMPLE_FMT_S16P: s->crossfade_samples = crossfade_samples_s16p; break;
case AV_SAMPLE_FMT_S32: s->crossfade_samples = crossfade_samples_s32; break;
case AV_SAMPLE_FMT_S32P: s->crossfade_samples = crossfade_samples_s32p; break;
}
config_output(outlink);
return 0;
}
static const AVFilterPad avfilter_af_acrossfade_outputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = acrossfade_config_output,
},
};
const FFFilter ff_af_acrossfade = {
.p.name = "acrossfade",
.p.description = NULL_IF_CONFIG_SMALL("Cross fade two input audio streams."),
.p.priv_class = &acrossfade_class,
.p.flags = AVFILTER_FLAG_DYNAMIC_INPUTS,
.priv_size = sizeof(AudioFadeContext),
.init = acrossfade_init,
.activate = activate,
FILTER_OUTPUTS(avfilter_af_acrossfade_outputs),
FILTER_SAMPLEFMTS_ARRAY(sample_fmts),
};
#endif /* CONFIG_ACROSSFADE_FILTER */
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