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FFmpeg/libavfilter/af_afade.c
Andreas Rheinhardt b4f5201967 avfilter: Replace query_formats callback with union of list and callback 
If one looks at the many query_formats callbacks in existence,
one will immediately recognize that there is one type of default
callback for video and a slightly different default callback for
audio: It is "return ff_set_common_formats_from_list(ctx, pix_fmts);"
for video with a filter-specific pix_fmts list. For audio, it is
the same with a filter-specific sample_fmts list together with
ff_set_common_all_samplerates() and ff_set_common_all_channel_counts().

This commit allows to remove the boilerplate query_formats callbacks
by replacing said callback with a union consisting the old callback
and pointers for pixel and sample format arrays. For the not uncommon
case in which these lists only contain a single entry (besides the
sentinel) enum AVPixelFormat and enum AVSampleFormat fields are also
added to the union to store them directly in the AVFilter,
thereby avoiding a relocation.

The state of said union will be contained in a new, dedicated AVFilter
field (the nb_inputs and nb_outputs fields have been shrunk to uint8_t
in order to create a hole for this new field; this is no problem, as
the maximum of all the nb_inputs is four; for nb_outputs it is only
two).

The state's default value coincides with the earlier default of
query_formats being unset, namely that the filter accepts all formats
(and also sample rates and channel counts/layouts for audio)
provided that these properties agree coincide for all inputs and
outputs.

By using different union members for audio and video filters
the type-unsafety of using the same functions for audio and video
lists will furthermore be more confined to formats.c than before.

When the new fields are used, they will also avoid allocations:
Currently something nearly equivalent to ff_default_query_formats()
is called after every successful call to a query_formats callback;
yet in the common case that the newly allocated AVFilterFormats
are not used at all (namely if there are no free links) these newly
allocated AVFilterFormats are freed again without ever being used.
Filters no longer using the callback will not exhibit this any more.

Reviewed-by: Paul B Mahol <onemda@gmail.com>
Reviewed-by: Nicolas George <george@nsup.org>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2021-10-05 17:48:25 +02:00

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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 "libavutil/opt.h"
#include "audio.h"
#include "avfilter.h"
#include "filters.h"
#include "internal.h"
typedef struct AudioFadeContext {
const AVClass *class;
int type;
int curve, curve2;
int64_t nb_samples;
int64_t start_sample;
int64_t duration;
int64_t start_time;
int overlap;
int cf0_eof;
int crossfade_is_over;
int64_t pts;
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);
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, 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 int query_formats(AVFilterContext *ctx)
{
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
};
int ret = ff_set_common_all_channel_counts(ctx);
if (ret < 0)
return ret;
ret = ff_set_common_formats_from_list(ctx, sample_fmts);
if (ret < 0)
return ret;
return ff_set_common_all_samplerates(ctx);
}
static double fade_gain(int curve, int64_t index, int64_t range)
{
#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 NONE:
gain = 1.0;
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, int64_t 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, int64_t 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 *s = ctx->priv;
switch (outlink->format) {
case AV_SAMPLE_FMT_DBL: s->fade_samples = fade_samples_dbl; break;
case AV_SAMPLE_FMT_DBLP: s->fade_samples = fade_samples_dblp; break;
case AV_SAMPLE_FMT_FLT: s->fade_samples = fade_samples_flt; break;
case AV_SAMPLE_FMT_FLTP: s->fade_samples = fade_samples_fltp; break;
case AV_SAMPLE_FMT_S16: s->fade_samples = fade_samples_s16; break;
case AV_SAMPLE_FMT_S16P: s->fade_samples = fade_samples_s16p; break;
case AV_SAMPLE_FMT_S32: s->fade_samples = fade_samples_s32; break;
case AV_SAMPLE_FMT_S32P: s->fade_samples = fade_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, "type" },
{ "t", "set the fade direction", OFFSET(type), AV_OPT_TYPE_INT, {.i64 = 0 }, 0, 1, TFLAGS, "type" },
{ "in", "fade-in", 0, AV_OPT_TYPE_CONST, {.i64 = 0 }, 0, 0, TFLAGS, "type" },
{ "out", "fade-out", 0, AV_OPT_TYPE_CONST, {.i64 = 1 }, 0, 0, TFLAGS, "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, "curve" },
{ "c", "set fade curve type", OFFSET(curve), AV_OPT_TYPE_INT, {.i64 = TRI }, NONE, NB_CURVES - 1, TFLAGS, "curve" },
{ "nofade", "no fade; keep audio as-is", 0, AV_OPT_TYPE_CONST, {.i64 = NONE }, 0, 0, TFLAGS, "curve" },
{ "tri", "linear slope", 0, AV_OPT_TYPE_CONST, {.i64 = TRI }, 0, 0, TFLAGS, "curve" },
{ "qsin", "quarter of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = QSIN }, 0, 0, TFLAGS, "curve" },
{ "esin", "exponential sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = ESIN }, 0, 0, TFLAGS, "curve" },
{ "hsin", "half of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = HSIN }, 0, 0, TFLAGS, "curve" },
{ "log", "logarithmic", 0, AV_OPT_TYPE_CONST, {.i64 = LOG }, 0, 0, TFLAGS, "curve" },
{ "ipar", "inverted parabola", 0, AV_OPT_TYPE_CONST, {.i64 = IPAR }, 0, 0, TFLAGS, "curve" },
{ "qua", "quadratic", 0, AV_OPT_TYPE_CONST, {.i64 = QUA }, 0, 0, TFLAGS, "curve" },
{ "cub", "cubic", 0, AV_OPT_TYPE_CONST, {.i64 = CUB }, 0, 0, TFLAGS, "curve" },
{ "squ", "square root", 0, AV_OPT_TYPE_CONST, {.i64 = SQU }, 0, 0, TFLAGS, "curve" },
{ "cbr", "cubic root", 0, AV_OPT_TYPE_CONST, {.i64 = CBR }, 0, 0, TFLAGS, "curve" },
{ "par", "parabola", 0, AV_OPT_TYPE_CONST, {.i64 = PAR }, 0, 0, TFLAGS, "curve" },
{ "exp", "exponential", 0, AV_OPT_TYPE_CONST, {.i64 = EXP }, 0, 0, TFLAGS, "curve" },
{ "iqsin", "inverted quarter of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = IQSIN}, 0, 0, TFLAGS, "curve" },
{ "ihsin", "inverted half of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = IHSIN}, 0, 0, TFLAGS, "curve" },
{ "dese", "double-exponential seat", 0, AV_OPT_TYPE_CONST, {.i64 = DESE }, 0, 0, TFLAGS, "curve" },
{ "desi", "double-exponential sigmoid", 0, AV_OPT_TYPE_CONST, {.i64 = DESI }, 0, 0, TFLAGS, "curve" },
{ "losi", "logistic sigmoid", 0, AV_OPT_TYPE_CONST, {.i64 = LOSI }, 0, 0, TFLAGS, "curve" },
{ "sinc", "sine cardinal function", 0, AV_OPT_TYPE_CONST, {.i64 = SINC }, 0, 0, TFLAGS, "curve" },
{ "isinc", "inverted sine cardinal function", 0, AV_OPT_TYPE_CONST, {.i64 = ISINC}, 0, 0, TFLAGS, "curve" },
{ 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->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))) {
av_samples_set_silence(out_buf->extended_data, 0, nb_samples,
out_buf->channels, out_buf->format);
} 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->channels,
s->type ? -1 : 1, start,
s->nb_samples, s->curve);
}
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 AVFilter ff_af_afade = {
.name = "afade",
.description = NULL_IF_CONFIG_SMALL("Fade in/out input audio."),
.priv_size = sizeof(AudioFadeContext),
.init = init,
FILTER_INPUTS(avfilter_af_afade_inputs),
FILTER_OUTPUTS(avfilter_af_afade_outputs),
FILTER_QUERY_FUNC(query_formats),
.priv_class = &afade_class,
.process_command = process_command,
.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC,
};
#endif /* CONFIG_AFADE_FILTER */
#if CONFIG_ACROSSFADE_FILTER
static const AVOption acrossfade_options[] = {
{ "nb_samples", "set number of samples for cross fade duration", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64 = 44100}, 1, INT32_MAX/10, FLAGS },
{ "ns", "set number of samples for cross fade duration", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.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, "curve" },
{ "c1", "set fade curve type for 1st stream", OFFSET(curve), AV_OPT_TYPE_INT, {.i64 = TRI }, NONE, NB_CURVES - 1, FLAGS, "curve" },
{ "nofade", "no fade; keep audio as-is", 0, AV_OPT_TYPE_CONST, {.i64 = NONE }, 0, 0, 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" },
{ "ipar", "inverted parabola", 0, AV_OPT_TYPE_CONST, {.i64 = IPAR }, 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" },
{ "par", "parabola", 0, AV_OPT_TYPE_CONST, {.i64 = PAR }, 0, 0, FLAGS, "curve" },
{ "exp", "exponential", 0, AV_OPT_TYPE_CONST, {.i64 = EXP }, 0, 0, FLAGS, "curve" },
{ "iqsin", "inverted quarter of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = IQSIN}, 0, 0, FLAGS, "curve" },
{ "ihsin", "inverted half of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = IHSIN}, 0, 0, FLAGS, "curve" },
{ "dese", "double-exponential seat", 0, AV_OPT_TYPE_CONST, {.i64 = DESE }, 0, 0, FLAGS, "curve" },
{ "desi", "double-exponential sigmoid", 0, AV_OPT_TYPE_CONST, {.i64 = DESI }, 0, 0, FLAGS, "curve" },
{ "losi", "logistic sigmoid", 0, AV_OPT_TYPE_CONST, {.i64 = LOSI }, 0, 0, FLAGS, "curve" },
{ "sinc", "sine cardinal function", 0, AV_OPT_TYPE_CONST, {.i64 = SINC }, 0, 0, FLAGS, "curve" },
{ "isinc", "inverted sine cardinal function", 0, AV_OPT_TYPE_CONST, {.i64 = ISINC}, 0, 0, FLAGS, "curve" },
{ "curve2", "set fade curve type for 2nd stream", OFFSET(curve2), AV_OPT_TYPE_INT, {.i64 = TRI }, NONE, NB_CURVES - 1, FLAGS, "curve" },
{ "c2", "set fade curve type for 2nd stream", OFFSET(curve2), AV_OPT_TYPE_INT, {.i64 = TRI }, NONE, NB_CURVES - 1, FLAGS, "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); \
double gain1 = fade_gain(curve1, i, nb_samples); \
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); \
double gain1 = fade_gain(curve1, i, nb_samples); \
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 activate(AVFilterContext *ctx)
{
AudioFadeContext *s = ctx->priv;
AVFilterLink *outlink = ctx->outputs[0];
AVFrame *in = NULL, *out, *cf[2] = { NULL };
int ret = 0, nb_samples, status;
int64_t pts;
FF_FILTER_FORWARD_STATUS_BACK_ALL(outlink, ctx);
if (s->crossfade_is_over) {
ret = ff_inlink_consume_frame(ctx->inputs[1], &in);
if (ret > 0) {
in->pts = s->pts;
s->pts += av_rescale_q(in->nb_samples,
(AVRational){ 1, outlink->sample_rate }, outlink->time_base);
return ff_filter_frame(outlink, in);
} else if (ret < 0) {
return ret;
} else if (ff_inlink_acknowledge_status(ctx->inputs[1], &status, &pts)) {
ff_outlink_set_status(ctx->outputs[0], status, pts);
return 0;
} else if (!ret) {
if (ff_outlink_frame_wanted(ctx->outputs[0])) {
ff_inlink_request_frame(ctx->inputs[1]);
return 0;
}
}
}
nb_samples = ff_inlink_queued_samples(ctx->inputs[0]);
if (nb_samples > s->nb_samples) {
nb_samples -= s->nb_samples;
ret = ff_inlink_consume_samples(ctx->inputs[0], nb_samples, nb_samples, &in);
if (ret < 0)
return ret;
in->pts = s->pts;
s->pts += av_rescale_q(in->nb_samples,
(AVRational){ 1, outlink->sample_rate }, outlink->time_base);
return ff_filter_frame(outlink, in);
} else if (s->cf0_eof && nb_samples >= s->nb_samples &&
ff_inlink_queued_samples(ctx->inputs[1]) >= s->nb_samples) {
if (s->overlap) {
out = ff_get_audio_buffer(outlink, s->nb_samples);
if (!out)
return AVERROR(ENOMEM);
ret = ff_inlink_consume_samples(ctx->inputs[0], s->nb_samples, s->nb_samples, &cf[0]);
if (ret < 0) {
av_frame_free(&out);
return ret;
}
ret = ff_inlink_consume_samples(ctx->inputs[1], s->nb_samples, s->nb_samples, &cf[1]);
if (ret < 0) {
av_frame_free(&out);
return ret;
}
s->crossfade_samples(out->extended_data, cf[0]->extended_data,
cf[1]->extended_data,
s->nb_samples, out->channels,
s->curve, s->curve2);
out->pts = s->pts;
s->pts += av_rescale_q(s->nb_samples,
(AVRational){ 1, outlink->sample_rate }, outlink->time_base);
s->crossfade_is_over = 1;
av_frame_free(&cf[0]);
av_frame_free(&cf[1]);
return ff_filter_frame(outlink, out);
} else {
out = ff_get_audio_buffer(outlink, s->nb_samples);
if (!out)
return AVERROR(ENOMEM);
ret = ff_inlink_consume_samples(ctx->inputs[0], s->nb_samples, s->nb_samples, &cf[0]);
if (ret < 0) {
av_frame_free(&out);
return ret;
}
s->fade_samples(out->extended_data, cf[0]->extended_data, s->nb_samples,
outlink->channels, -1, s->nb_samples - 1, s->nb_samples, s->curve);
out->pts = s->pts;
s->pts += av_rescale_q(s->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;
out = ff_get_audio_buffer(outlink, s->nb_samples);
if (!out)
return AVERROR(ENOMEM);
ret = ff_inlink_consume_samples(ctx->inputs[1], s->nb_samples, s->nb_samples, &cf[1]);
if (ret < 0) {
av_frame_free(&out);
return ret;
}
s->fade_samples(out->extended_data, cf[1]->extended_data, s->nb_samples,
outlink->channels, 1, 0, s->nb_samples, s->curve2);
out->pts = s->pts;
s->pts += av_rescale_q(s->nb_samples,
(AVRational){ 1, outlink->sample_rate }, outlink->time_base);
s->crossfade_is_over = 1;
av_frame_free(&cf[1]);
return ff_filter_frame(outlink, out);
}
} else if (ff_outlink_frame_wanted(ctx->outputs[0])) {
if (!s->cf0_eof && ff_outlink_get_status(ctx->inputs[0])) {
s->cf0_eof = 1;
}
if (ff_outlink_get_status(ctx->inputs[1])) {
ff_outlink_set_status(ctx->outputs[0], AVERROR_EOF, AV_NOPTS_VALUE);
return 0;
}
if (!s->cf0_eof)
ff_inlink_request_frame(ctx->inputs[0]);
else
ff_inlink_request_frame(ctx->inputs[1]);
return 0;
}
return ret;
}
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_inputs[] = {
{
.name = "crossfade0",
.type = AVMEDIA_TYPE_AUDIO,
},
{
.name = "crossfade1",
.type = AVMEDIA_TYPE_AUDIO,
},
};
static const AVFilterPad avfilter_af_acrossfade_outputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = acrossfade_config_output,
},
};
const AVFilter ff_af_acrossfade = {
.name = "acrossfade",
.description = NULL_IF_CONFIG_SMALL("Cross fade two input audio streams."),
.priv_size = sizeof(AudioFadeContext),
.activate = activate,
.priv_class = &acrossfade_class,
FILTER_INPUTS(avfilter_af_acrossfade_inputs),
FILTER_OUTPUTS(avfilter_af_acrossfade_outputs),
FILTER_QUERY_FUNC(query_formats),
};
#endif /* CONFIG_ACROSSFADE_FILTER */
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