/*
* 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/audio_fifo.h"
#include "libavutil/opt.h"
#include "audio.h"
#include "avfilter.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;
AVAudioFifo *fifo[2];
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 { TRI, QSIN, ESIN, HSIN, LOG, IPAR, QUA, CUB, SQU, CBR, PAR, EXP, IQSIN, IHSIN, DESE, DESI, NB_CURVES };
#define OFFSET(x) offsetof(AudioFadeContext, x)
#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
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
};
int ret;
layouts = ff_all_channel_counts();
if (!layouts)
return AVERROR(ENOMEM);
ret = ff_set_common_channel_layouts(ctx, layouts);
if (ret < 0)
return ret;
formats = ff_make_format_list(sample_fmts);
if (!formats)
return AVERROR(ENOMEM);
ret = ff_set_common_formats(ctx, formats);
if (ret < 0)
return ret;
formats = ff_all_samplerates();
if (!formats)
return AVERROR(ENOMEM);
return ff_set_common_samplerates(ctx, formats);
}
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;
}
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);
if (s->start_time)
s->start_sample = av_rescale(s->start_time, outlink->sample_rate, AV_TIME_BASE);
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, FLAGS, "type" },
{ "t", "set the fade direction", OFFSET(type), AV_OPT_TYPE_INT, {.i64 = 0 }, 0, 1, FLAGS, "type" },
{ "in", "fade-in", 0, AV_OPT_TYPE_CONST, {.i64 = 0 }, 0, 0, FLAGS, "type" },
{ "out", "fade-out", 0, AV_OPT_TYPE_CONST, {.i64 = 1 }, 0, 0, FLAGS, "type" },
{ "start_sample", "set number of first sample to start fading", OFFSET(start_sample), AV_OPT_TYPE_INT64, {.i64 = 0 }, 0, INT64_MAX, FLAGS },
{ "ss", "set number of first sample to start fading", OFFSET(start_sample), AV_OPT_TYPE_INT64, {.i64 = 0 }, 0, INT64_MAX, FLAGS },
{ "nb_samples", "set number of samples for fade duration", OFFSET(nb_samples), AV_OPT_TYPE_INT64, {.i64 = 44100}, 1, INT64_MAX, FLAGS },
{ "ns", "set number of samples for fade duration", OFFSET(nb_samples), AV_OPT_TYPE_INT64, {.i64 = 44100}, 1, INT64_MAX, FLAGS },
{ "start_time", "set time to start fading", OFFSET(start_time), AV_OPT_TYPE_DURATION, {.i64 = 0. }, 0, INT64_MAX, FLAGS },
{ "st", "set time to start fading", OFFSET(start_time), AV_OPT_TYPE_DURATION, {.i64 = 0. }, 0, INT64_MAX, FLAGS },
{ "duration", "set fade duration", OFFSET(duration), AV_OPT_TYPE_DURATION, {.i64 = 0. }, 0, INT32_MAX, FLAGS },
{ "d", "set fade duration", OFFSET(duration), AV_OPT_TYPE_DURATION, {.i64 = 0. }, 0, INT32_MAX, FLAGS },
{ "curve", "set fade curve type", OFFSET(curve), AV_OPT_TYPE_INT, {.i64 = TRI }, 0, NB_CURVES - 1, FLAGS, "curve" },
{ "c", "set fade curve type", OFFSET(curve), AV_OPT_TYPE_INT, {.i64 = TRI }, 0, NB_CURVES - 1, 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" },
{ 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(inlink, 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 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 ff_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,
.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, 60, FLAGS },
{ "d", "set cross fade duration", OFFSET(duration), AV_OPT_TYPE_DURATION, {.i64 = 0. }, 0, 60, 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 }, 0, NB_CURVES - 1, FLAGS, "curve" },
{ "c1", "set fade curve type for 1st stream", OFFSET(curve), AV_OPT_TYPE_INT, {.i64 = TRI }, 0, NB_CURVES - 1, 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" },
{ "curve2", "set fade curve type for 2nd stream", OFFSET(curve2), AV_OPT_TYPE_INT, {.i64 = TRI }, 0, NB_CURVES - 1, FLAGS, "curve" },
{ "c2", "set fade curve type for 2nd stream", OFFSET(curve2), AV_OPT_TYPE_INT, {.i64 = TRI }, 0, 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 acrossfade_filter_frame(AVFilterLink *inlink, AVFrame *in)
{
AVFilterContext *ctx = inlink->dst;
AudioFadeContext *s = ctx->priv;
AVFilterLink *outlink = ctx->outputs[0];
AVFrame *out, *cf[2] = { NULL };
int ret = 0, nb_samples;
if (s->crossfade_is_over) {
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 (inlink == ctx->inputs[0]) {
av_audio_fifo_write(s->fifo[0], (void **)in->extended_data, in->nb_samples);
nb_samples = av_audio_fifo_size(s->fifo[0]) - s->nb_samples;
if (nb_samples > 0) {
out = ff_get_audio_buffer(outlink, nb_samples);
if (!out) {
ret = AVERROR(ENOMEM);
goto fail;
}
av_audio_fifo_read(s->fifo[0], (void **)out->extended_data, nb_samples);
out->pts = s->pts;
s->pts += av_rescale_q(nb_samples,
(AVRational){ 1, outlink->sample_rate }, outlink->time_base);
ret = ff_filter_frame(outlink, out);
}
} else if (av_audio_fifo_size(s->fifo[1]) < s->nb_samples) {
if (!s->overlap && av_audio_fifo_size(s->fifo[0]) > 0) {
nb_samples = av_audio_fifo_size(s->fifo[0]);
cf[0] = ff_get_audio_buffer(outlink, nb_samples);
out = ff_get_audio_buffer(outlink, nb_samples);
if (!out || !cf[0]) {
ret = AVERROR(ENOMEM);
goto fail;
}
av_audio_fifo_read(s->fifo[0], (void **)cf[0]->extended_data, nb_samples);
s->fade_samples(out->extended_data, cf[0]->extended_data, nb_samples,
outlink->channels, -1, nb_samples - 1, nb_samples, s->curve);
out->pts = s->pts;
s->pts += av_rescale_q(nb_samples,
(AVRational){ 1, outlink->sample_rate }, outlink->time_base);
ret = ff_filter_frame(outlink, out);
if (ret < 0)
goto fail;
}
av_audio_fifo_write(s->fifo[1], (void **)in->extended_data, in->nb_samples);
} else if (av_audio_fifo_size(s->fifo[1]) >= s->nb_samples) {
av_audio_fifo_write(s->fifo[1], (void **)in->extended_data, in->nb_samples);
if (s->overlap) {
cf[0] = ff_get_audio_buffer(outlink, s->nb_samples);
cf[1] = ff_get_audio_buffer(outlink, s->nb_samples);
out = ff_get_audio_buffer(outlink, s->nb_samples);
if (!out || !cf[0] || !cf[1]) {
av_frame_free(&out);
ret = AVERROR(ENOMEM);
goto fail;
}
av_audio_fifo_read(s->fifo[0], (void **)cf[0]->extended_data, s->nb_samples);
av_audio_fifo_read(s->fifo[1], (void **)cf[1]->extended_data, s->nb_samples);
s->crossfade_samples(out->extended_data, cf[0]->extended_data,
cf[1]->extended_data,
s->nb_samples, in->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);
ret = ff_filter_frame(outlink, out);
if (ret < 0)
goto fail;
} else {
out = ff_get_audio_buffer(outlink, s->nb_samples);
cf[1] = ff_get_audio_buffer(outlink, s->nb_samples);
if (!out || !cf[1]) {
ret = AVERROR(ENOMEM);
av_frame_free(&out);
goto fail;
}
av_audio_fifo_read(s->fifo[1], (void **)cf[1]->extended_data, s->nb_samples);
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);
ret = ff_filter_frame(outlink, out);
if (ret < 0)
goto fail;
}
nb_samples = av_audio_fifo_size(s->fifo[1]);
if (nb_samples > 0) {
out = ff_get_audio_buffer(outlink, nb_samples);
if (!out) {
ret = AVERROR(ENOMEM);
goto fail;
}
av_audio_fifo_read(s->fifo[1], (void **)out->extended_data, nb_samples);
out->pts = s->pts;
s->pts += av_rescale_q(nb_samples,
(AVRational){ 1, outlink->sample_rate }, outlink->time_base);
ret = ff_filter_frame(outlink, out);
}
s->crossfade_is_over = 1;
}
fail:
av_frame_free(&in);
av_frame_free(&cf[0]);
av_frame_free(&cf[1]);
return ret;
}
static int acrossfade_request_frame(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
AudioFadeContext *s = ctx->priv;
int ret = 0;
if (!s->cf0_eof) {
AVFilterLink *cf0 = ctx->inputs[0];
ret = ff_request_frame(cf0);
if (ret < 0 && ret != AVERROR_EOF)
return ret;
if (ret == AVERROR_EOF) {
s->cf0_eof = 1;
ret = 0;
}
} else {
AVFilterLink *cf1 = ctx->inputs[1];
int nb_samples = av_audio_fifo_size(s->fifo[1]);
ret = ff_request_frame(cf1);
if (ret == AVERROR_EOF && nb_samples > 0) {
AVFrame *out = ff_get_audio_buffer(outlink, nb_samples);
if (!out)
return AVERROR(ENOMEM);
av_audio_fifo_read(s->fifo[1], (void **)out->extended_data, nb_samples);
ret = ff_filter_frame(outlink, out);
}
}
return ret;
}
static int acrossfade_config_output(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
AudioFadeContext *s = ctx->priv;
if (ctx->inputs[0]->sample_rate != ctx->inputs[1]->sample_rate) {
av_log(ctx, AV_LOG_ERROR,
"Inputs must have the same sample rate "
"%d for in0 vs %d for in1\n",
ctx->inputs[0]->sample_rate, ctx->inputs[1]->sample_rate);
return AVERROR(EINVAL);
}
outlink->sample_rate = ctx->inputs[0]->sample_rate;
outlink->time_base = ctx->inputs[0]->time_base;
outlink->channel_layout = ctx->inputs[0]->channel_layout;
outlink->channels = ctx->inputs[0]->channels;
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);
s->fifo[0] = av_audio_fifo_alloc(outlink->format, outlink->channels, s->nb_samples);
s->fifo[1] = av_audio_fifo_alloc(outlink->format, outlink->channels, s->nb_samples);
if (!s->fifo[0] || !s->fifo[1])
return AVERROR(ENOMEM);
return 0;
}
static av_cold void uninit(AVFilterContext *ctx)
{
AudioFadeContext *s = ctx->priv;
av_audio_fifo_free(s->fifo[0]);
av_audio_fifo_free(s->fifo[1]);
}
static const AVFilterPad avfilter_af_acrossfade_inputs[] = {
{
.name = "crossfade0",
.type = AVMEDIA_TYPE_AUDIO,
.filter_frame = acrossfade_filter_frame,
},
{
.name = "crossfade1",
.type = AVMEDIA_TYPE_AUDIO,
.filter_frame = acrossfade_filter_frame,
},
{ NULL }
};
static const AVFilterPad avfilter_af_acrossfade_outputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.request_frame = acrossfade_request_frame,
.config_props = acrossfade_config_output,
},
{ NULL }
};
AVFilter ff_af_acrossfade = {
.name = "acrossfade",
.description = NULL_IF_CONFIG_SMALL("Cross fade two input audio streams."),
.query_formats = query_formats,
.priv_size = sizeof(AudioFadeContext),
.uninit = uninit,
.priv_class = &acrossfade_class,
.inputs = avfilter_af_acrossfade_inputs,
.outputs = avfilter_af_acrossfade_outputs,
};
#endif /* CONFIG_ACROSSFADE_FILTER */