[deb_ffmpeg.git] / ffmpeg / libavfilter / af_afade.c

/*
 * Copyright (c) 2013 Paul B Mahol
 *
 * This file is part of FFmpeg.
 *
 * FFmpeg is free software; you can redistribute it and/or
 * modify it under the terms of the GNU Lesser General Public
 * License as published by the Free Software Foundation; either
 * version 2.1 of the License, or (at your option) any later version.
 *
 * FFmpeg is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
 * Lesser General Public License for more details.
 *
 * You should have received a copy of the GNU Lesser General Public
 * License along with FFmpeg; if not, write to the Free Software
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */

/**
 * @file
 * fade audio filter
 */

#include "libavutil/opt.h"
#include "audio.h"
#include "avfilter.h"
#include "internal.h"

typedef struct {
    const AVClass *class;
    int type;
    int curve;
    int nb_samples;
    int64_t start_sample;
    int64_t duration;
    int64_t start_time;

    void (*fade_samples)(uint8_t **dst, uint8_t * const *src,
                         int nb_samples, int channels, int direction,
                         int64_t start, int range, int curve);
} AudioFadeContext;

enum CurveType { TRI, QSIN, ESIN, HSIN, LOG, PAR, QUA, CUB, SQU, CBR };

#define OFFSET(x) offsetof(AudioFadeContext, x)
#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM

static const AVOption afade_options[] = {
    { "type",         "set the fade direction",                      OFFSET(type),         AV_OPT_TYPE_INT,    {.i64 = 0    }, 0, 1, FLAGS, "type" },
    { "t",            "set the fade direction",                      OFFSET(type),         AV_OPT_TYPE_INT,    {.i64 = 0    }, 0, 1, FLAGS, "type" },
    { "in",           "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_INT,    {.i64 = 44100}, 1, INT32_MAX, FLAGS },
    { "ns",           "set number of samples for fade duration",     OFFSET(nb_samples),   AV_OPT_TYPE_INT,    {.i64 = 44100}, 1, INT32_MAX, FLAGS },
    { "start_time",   "set time to start fading",                    OFFSET(start_time),   AV_OPT_TYPE_DURATION, {.i64 = 0. }, 0, INT32_MAX, FLAGS },
    { "st",           "set time to start fading",                    OFFSET(start_time),   AV_OPT_TYPE_DURATION, {.i64 = 0. }, 0, INT32_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  }, TRI, CBR, FLAGS, "curve" },
    { "c",            "set fade curve type",                         OFFSET(curve),        AV_OPT_TYPE_INT,    {.i64 = TRI  }, TRI, CBR, FLAGS, "curve" },
    { "tri",          "linear slope",                                0,                    AV_OPT_TYPE_CONST,  {.i64 = TRI  }, 0, 0, FLAGS, "curve" },
    { "qsin",         "quarter of sine wave",                        0,                    AV_OPT_TYPE_CONST,  {.i64 = QSIN }, 0, 0, FLAGS, "curve" },
    { "esin",         "exponential sine wave",                       0,                    AV_OPT_TYPE_CONST,  {.i64 = ESIN }, 0, 0, FLAGS, "curve" },
    { "hsin",         "half of sine wave",                           0,                    AV_OPT_TYPE_CONST,  {.i64 = HSIN }, 0, 0, FLAGS, "curve" },
    { "log",          "logarithmic",                                 0,                    AV_OPT_TYPE_CONST,  {.i64 = LOG  }, 0, 0, FLAGS, "curve" },
    { "par",          "inverted parabola",                           0,                    AV_OPT_TYPE_CONST,  {.i64 = PAR  }, 0, 0, FLAGS, "curve" },
    { "qua",          "quadratic",                                   0,                    AV_OPT_TYPE_CONST,  {.i64 = QUA  }, 0, 0, FLAGS, "curve" },
    { "cub",          "cubic",                                       0,                    AV_OPT_TYPE_CONST,  {.i64 = CUB  }, 0, 0, FLAGS, "curve" },
    { "squ",          "square root",                                 0,                    AV_OPT_TYPE_CONST,  {.i64 = SQU  }, 0, 0, FLAGS, "curve" },
    { "cbr",          "cubic root",                                  0,                    AV_OPT_TYPE_CONST,  {.i64 = CBR  }, 0, 0, FLAGS, "curve" },
    { NULL }
};

AVFILTER_DEFINE_CLASS(afade);

static av_cold int init(AVFilterContext *ctx)
{
    AudioFadeContext *s = ctx->priv;

    if (INT64_MAX - s->nb_samples < s->start_sample)
        return AVERROR(EINVAL);

    return 0;
}

static int query_formats(AVFilterContext *ctx)
{
    AVFilterFormats *formats;
    AVFilterChannelLayouts *layouts;
    static const enum AVSampleFormat sample_fmts[] = {
        AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16P,
        AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32P,
        AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP,
        AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBLP,
        AV_SAMPLE_FMT_NONE
    };

    layouts = ff_all_channel_layouts();
    if (!layouts)
        return AVERROR(ENOMEM);
    ff_set_common_channel_layouts(ctx, layouts);

    formats = ff_make_format_list(sample_fmts);
    if (!formats)
        return AVERROR(ENOMEM);
    ff_set_common_formats(ctx, formats);

    formats = ff_all_samplerates();
    if (!formats)
        return AVERROR(ENOMEM);
    ff_set_common_samplerates(ctx, formats);

    return 0;
}

static double fade_gain(int curve, int64_t index, int range)
{
    double gain;

    gain = FFMAX(0.0, FFMIN(1.0, 1.0 * index / range));

    switch (curve) {
    case QSIN:
        gain = sin(gain * M_PI / 2.0);
        break;
    case ESIN:
        gain = 1.0 - cos(M_PI / 4.0 * (pow(2.0*gain - 1, 3) + 1));
        break;
    case HSIN:
        gain = (1.0 - cos(gain * M_PI)) / 2.0;
        break;
    case LOG:
        gain = pow(0.1, (1 - gain) * 5.0);
        break;
    case PAR:
        gain = (1 - (1 - gain) * (1 - gain));
        break;
    case QUA:
        gain *= gain;
        break;
    case CUB:
        gain = gain * gain * gain;
        break;
    case SQU:
        gain = sqrt(gain);
        break;
    case CBR:
        gain = cbrt(gain);
        break;
    }

    return gain;
}

#define FADE_PLANAR(name, type)                                             \
static void fade_samples_## name ##p(uint8_t **dst, uint8_t * const *src,   \
                                     int nb_samples, int channels, int dir, \
                                     int64_t start, int range, int curve)   \
{                                                                           \
    int i, c;                                                               \
                                                                            \
    for (i = 0; i < nb_samples; i++) {                                      \
        double gain = fade_gain(curve, start + i * dir, range);             \
        for (c = 0; c < channels; c++) {                                    \
            type *d = (type *)dst[c];                                       \
            const type *s = (type *)src[c];                                 \
                                                                            \
            d[i] = s[i] * gain;                                             \
        }                                                                   \
    }                                                                       \
}

#define FADE(name, type)                                                    \
static void fade_samples_## name (uint8_t **dst, uint8_t * const *src,      \
                                  int nb_samples, int channels, int dir,    \
                                  int64_t start, int range, int curve)      \
{                                                                           \
    type *d = (type *)dst[0];                                               \
    const type *s = (type *)src[0];                                         \
    int i, c, k = 0;                                                        \
                                                                            \
    for (i = 0; i < nb_samples; i++) {                                      \
        double gain = fade_gain(curve, start + i * dir, range);             \
        for (c = 0; c < channels; c++, k++)                                 \
            d[k] = s[k] * gain;                                             \
    }                                                                       \
}

FADE_PLANAR(dbl, double)
FADE_PLANAR(flt, float)
FADE_PLANAR(s16, int16_t)
FADE_PLANAR(s32, int32_t)

FADE(dbl, double)
FADE(flt, float)
FADE(s16, int16_t)
FADE(s32, int32_t)

static int config_input(AVFilterLink *inlink)
{
    AVFilterContext *ctx = inlink->dst;
    AudioFadeContext *s  = ctx->priv;

    switch (inlink->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, inlink->sample_rate, AV_TIME_BASE);
    if (s->start_time)
        s->start_sample = av_rescale(s->start_time, inlink->sample_rate, AV_TIME_BASE);

    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, (AVRational){1, outlink->sample_rate}, outlink->time_base);

    if ((!s->type && (s->start_sample + s->nb_samples < cur_sample)) ||
        ( s->type && (cur_sample + s->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,
                               av_frame_get_channels(out_buf), 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, av_frame_get_channels(buf),
                        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,
        .config_props = config_input,
    },
    { NULL }
};

static const AVFilterPad avfilter_af_afade_outputs[] = {
    {
        .name = "default",
        .type = AVMEDIA_TYPE_AUDIO,
    },
    { 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,
};
Commit	Line	Data
2ba45a60 DM	1	/*
	2	* Copyright (c) 2013 Paul B Mahol
	3	*
	4	* This file is part of FFmpeg.
	5	*
	6	* FFmpeg is free software; you can redistribute it and/or
	7	* modify it under the terms of the GNU Lesser General Public
	8	* License as published by the Free Software Foundation; either
	9	* version 2.1 of the License, or (at your option) any later version.
	10	*
	11	* FFmpeg is distributed in the hope that it will be useful,
	12	* but WITHOUT ANY WARRANTY; without even the implied warranty of
	13	* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
	14	* Lesser General Public License for more details.
	15	*
	16	* You should have received a copy of the GNU Lesser General Public
	17	* License along with FFmpeg; if not, write to the Free Software
	18	* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
	19	*/
	20
	21	/**
	22	* @file
	23	* fade audio filter
	24	*/
	25
	26	#include "libavutil/opt.h"
	27	#include "audio.h"
	28	#include "avfilter.h"
	29	#include "internal.h"
	30
	31	typedef struct {
	32	const AVClass *class;
	33	int type;
	34	int curve;
	35	int nb_samples;
	36	int64_t start_sample;
	37	int64_t duration;
	38	int64_t start_time;
	39
	40	void (fade_samples)(uint8_t dst, uint8_t const *src,
	41	int nb_samples, int channels, int direction,
	42	int64_t start, int range, int curve);
	43	} AudioFadeContext;
	44
	45	enum CurveType { TRI, QSIN, ESIN, HSIN, LOG, PAR, QUA, CUB, SQU, CBR };
	46
	47	#define OFFSET(x) offsetof(AudioFadeContext, x)
	48	#define FLAGS AV_OPT_FLAG_AUDIO_PARAM\|AV_OPT_FLAG_FILTERING_PARAM
	49
	50	static const AVOption afade_options[] = {
	51	{ "type", "set the fade direction", OFFSET(type), AV_OPT_TYPE_INT, {.i64 = 0 }, 0, 1, FLAGS, "type" },
	52	{ "t", "set the fade direction", OFFSET(type), AV_OPT_TYPE_INT, {.i64 = 0 }, 0, 1, FLAGS, "type" },
	53	{ "in", "fade-in", 0, AV_OPT_TYPE_CONST, {.i64 = 0 }, 0, 0, FLAGS, "type" },
	54	{ "out", "fade-out", 0, AV_OPT_TYPE_CONST, {.i64 = 1 }, 0, 0, FLAGS, "type" },
	55	{ "start_sample", "set number of first sample to start fading", OFFSET(start_sample), AV_OPT_TYPE_INT64, {.i64 = 0 }, 0, INT64_MAX, FLAGS },
	56	{ "ss", "set number of first sample to start fading", OFFSET(start_sample), AV_OPT_TYPE_INT64, {.i64 = 0 }, 0, INT64_MAX, FLAGS },
	57	{ "nb_samples", "set number of samples for fade duration", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64 = 44100}, 1, INT32_MAX, FLAGS },
	58	{ "ns", "set number of samples for fade duration", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64 = 44100}, 1, INT32_MAX, FLAGS },
	59	{ "start_time", "set time to start fading", OFFSET(start_time), AV_OPT_TYPE_DURATION, {.i64 = 0. }, 0, INT32_MAX, FLAGS },
	60	{ "st", "set time to start fading", OFFSET(start_time), AV_OPT_TYPE_DURATION, {.i64 = 0. }, 0, INT32_MAX, FLAGS },
	61	{ "duration", "set fade duration", OFFSET(duration), AV_OPT_TYPE_DURATION, {.i64 = 0. }, 0, INT32_MAX, FLAGS },
	62	{ "d", "set fade duration", OFFSET(duration), AV_OPT_TYPE_DURATION, {.i64 = 0. }, 0, INT32_MAX, FLAGS },
	63	{ "curve", "set fade curve type", OFFSET(curve), AV_OPT_TYPE_INT, {.i64 = TRI }, TRI, CBR, FLAGS, "curve" },
	64	{ "c", "set fade curve type", OFFSET(curve), AV_OPT_TYPE_INT, {.i64 = TRI }, TRI, CBR, FLAGS, "curve" },
65	{ "tri", "linear slope", 0, AV_OPT_TYPE_CONST, {.i64 = TRI }, 0, 0, FLAGS, "curve" },
66	{ "qsin", "quarter of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = QSIN }, 0, 0, FLAGS, "curve" },
67	{ "esin", "exponential sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = ESIN }, 0, 0, FLAGS, "curve" },
68	{ "hsin", "half of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = HSIN }, 0, 0, FLAGS, "curve" },
69	{ "log", "logarithmic", 0, AV_OPT_TYPE_CONST, {.i64 = LOG }, 0, 0, FLAGS, "curve" },
70	{ "par", "inverted parabola", 0, AV_OPT_TYPE_CONST, {.i64 = PAR }, 0, 0, FLAGS, "curve" },
71	{ "qua", "quadratic", 0, AV_OPT_TYPE_CONST, {.i64 = QUA }, 0, 0, FLAGS, "curve" },
72	{ "cub", "cubic", 0, AV_OPT_TYPE_CONST, {.i64 = CUB }, 0, 0, FLAGS, "curve" },
73	{ "squ", "square root", 0, AV_OPT_TYPE_CONST, {.i64 = SQU }, 0, 0, FLAGS, "curve" },
74	{ "cbr", "cubic root", 0, AV_OPT_TYPE_CONST, {.i64 = CBR }, 0, 0, FLAGS, "curve" },
75	{ NULL }
76	};
77
78	AVFILTER_DEFINE_CLASS(afade);
79
80	static av_cold int init(AVFilterContext *ctx)
81	{
82	AudioFadeContext *s = ctx->priv;
83
84	if (INT64_MAX - s->nb_samples < s->start_sample)
85	return AVERROR(EINVAL);
86
87	return 0;
88	}
89
90	static int query_formats(AVFilterContext *ctx)
91	{
92	AVFilterFormats *formats;
93	AVFilterChannelLayouts *layouts;
94	static const enum AVSampleFormat sample_fmts[] = {
95	AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16P,
96	AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32P,
97	AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP,
98	AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBLP,
99	AV_SAMPLE_FMT_NONE
100	};
101
102	layouts = ff_all_channel_layouts();
103	if (!layouts)
104	return AVERROR(ENOMEM);
105	ff_set_common_channel_layouts(ctx, layouts);
106
107	formats = ff_make_format_list(sample_fmts);
108	if (!formats)
109	return AVERROR(ENOMEM);
110	ff_set_common_formats(ctx, formats);
111
112	formats = ff_all_samplerates();
113	if (!formats)
114	return AVERROR(ENOMEM);
115	ff_set_common_samplerates(ctx, formats);
116
117	return 0;
118	}
119
120	static double fade_gain(int curve, int64_t index, int range)
121	{
122	double gain;
123
124	gain = FFMAX(0.0, FFMIN(1.0, 1.0 * index / range));
125
126	switch (curve) {
127	case QSIN:
128	gain = sin(gain * M_PI / 2.0);
129	break;
130	case ESIN:
131	gain = 1.0 - cos(M_PI / 4.0 * (pow(2.0*gain - 1, 3) + 1));
132	break;
133	case HSIN:
134	gain = (1.0 - cos(gain * M_PI)) / 2.0;
135	break;
136	case LOG:
137	gain = pow(0.1, (1 - gain) * 5.0);
138	break;
139	case PAR:
140	gain = (1 - (1 - gain) * (1 - gain));
141	break;
142	case QUA:
143	gain *= gain;
144	break;
145	case CUB:
146	gain = gain * gain * gain;
147	break;
148	case SQU:
149	gain = sqrt(gain);
150	break;
151	case CBR:
152	gain = cbrt(gain);
153	break;
154	}
155
156	return gain;
157	}
158
159	#define FADE_PLANAR(name, type) \
160	static void fade_samples_## name ##p(uint8_t *dst, uint8_t const *src, \
161	int nb_samples, int channels, int dir, \
162	int64_t start, int range, int curve) \
163	{ \
164	int i, c; \
165	\
166	for (i = 0; i < nb_samples; i++) { \
167	double gain = fade_gain(curve, start + i * dir, range); \
168	for (c = 0; c < channels; c++) { \
169	type d = (type )dst[c]; \
170	const type s = (type )src[c]; \
171	\
172	d[i] = s[i] * gain; \
173	} \
174	} \
175	}
176
177	#define FADE(name, type) \
178	static void fade_samples_## name (uint8_t *dst, uint8_t const *src, \
179	int nb_samples, int channels, int dir, \
180	int64_t start, int range, int curve) \
181	{ \
182	type d = (type )dst[0]; \
183	const type s = (type )src[0]; \
184	int i, c, k = 0; \
185	\
186	for (i = 0; i < nb_samples; i++) { \
187	double gain = fade_gain(curve, start + i * dir, range); \
188	for (c = 0; c < channels; c++, k++) \
189	d[k] = s[k] * gain; \
190	} \
191	}
192
193	FADE_PLANAR(dbl, double)
194	FADE_PLANAR(flt, float)
195	FADE_PLANAR(s16, int16_t)
196	FADE_PLANAR(s32, int32_t)
197
198	FADE(dbl, double)
199	FADE(flt, float)
200	FADE(s16, int16_t)
201	FADE(s32, int32_t)
202
203	static int config_input(AVFilterLink *inlink)
204	{
205	AVFilterContext *ctx = inlink->dst;
206	AudioFadeContext *s = ctx->priv;
207
208	switch (inlink->format) {
209	case AV_SAMPLE_FMT_DBL: s->fade_samples = fade_samples_dbl; break;
210	case AV_SAMPLE_FMT_DBLP: s->fade_samples = fade_samples_dblp; break;
211	case AV_SAMPLE_FMT_FLT: s->fade_samples = fade_samples_flt; break;
212	case AV_SAMPLE_FMT_FLTP: s->fade_samples = fade_samples_fltp; break;
213	case AV_SAMPLE_FMT_S16: s->fade_samples = fade_samples_s16; break;
214	case AV_SAMPLE_FMT_S16P: s->fade_samples = fade_samples_s16p; break;
215	case AV_SAMPLE_FMT_S32: s->fade_samples = fade_samples_s32; break;
216	case AV_SAMPLE_FMT_S32P: s->fade_samples = fade_samples_s32p; break;
217	}
218
219	if (s->duration)
220	s->nb_samples = av_rescale(s->duration, inlink->sample_rate, AV_TIME_BASE);
221	if (s->start_time)
222	s->start_sample = av_rescale(s->start_time, inlink->sample_rate, AV_TIME_BASE);
223
224	return 0;
225	}
226
227	static int filter_frame(AVFilterLink inlink, AVFrame buf)
228	{
229	AudioFadeContext *s = inlink->dst->priv;
230	AVFilterLink *outlink = inlink->dst->outputs[0];
231	int nb_samples = buf->nb_samples;
232	AVFrame *out_buf;
233	int64_t cur_sample = av_rescale_q(buf->pts, (AVRational){1, outlink->sample_rate}, outlink->time_base);
234
235	if ((!s->type && (s->start_sample + s->nb_samples < cur_sample)) \|\|
236	( s->type && (cur_sample + s->nb_samples < s->start_sample)))
237	return ff_filter_frame(outlink, buf);
238
239	if (av_frame_is_writable(buf)) {
240	out_buf = buf;
241	} else {
242	out_buf = ff_get_audio_buffer(inlink, nb_samples);
243	if (!out_buf)
244	return AVERROR(ENOMEM);
245	av_frame_copy_props(out_buf, buf);
246	}
247
248	if ((!s->type && (cur_sample + nb_samples < s->start_sample)) \|\|
249	( s->type && (s->start_sample + s->nb_samples < cur_sample))) {
250	av_samples_set_silence(out_buf->extended_data, 0, nb_samples,
251	av_frame_get_channels(out_buf), out_buf->format);
252	} else {
253	int64_t start;
254
255	if (!s->type)
256	start = cur_sample - s->start_sample;
257	else
258	start = s->start_sample + s->nb_samples - cur_sample;
259
260	s->fade_samples(out_buf->extended_data, buf->extended_data,
261	nb_samples, av_frame_get_channels(buf),
262	s->type ? -1 : 1, start,
263	s->nb_samples, s->curve);
264	}
265
266	if (buf != out_buf)
267	av_frame_free(&buf);
268
269	return ff_filter_frame(outlink, out_buf);
270	}
271
272	static const AVFilterPad avfilter_af_afade_inputs[] = {
273	{
274	.name = "default",
275	.type = AVMEDIA_TYPE_AUDIO,
276	.filter_frame = filter_frame,
277	.config_props = config_input,
278	},
279	{ NULL }
280	};
281
282	static const AVFilterPad avfilter_af_afade_outputs[] = {
283	{
284	.name = "default",
285	.type = AVMEDIA_TYPE_AUDIO,
286	},
287	{ NULL }
288	};
289
290	AVFilter ff_af_afade = {
291	.name = "afade",
292	.description = NULL_IF_CONFIG_SMALL("Fade in/out input audio."),
293	.query_formats = query_formats,
294	.priv_size = sizeof(AudioFadeContext),
295	.init = init,
296	.inputs = avfilter_af_afade_inputs,
297	.outputs = avfilter_af_afade_outputs,
298	.priv_class = &afade_class,
299	.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC,
300	};