WebSVN – Kolibri OS – Blame – /contrib/sdk/sources/ffmpeg/ffmpeg-2.1/libavfilter/af_afade.c

Rev	Author	Line No.	Line
6148	serge	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 avfilter_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	};

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(root)/contrib/sdk/sources/ffmpeg/ffmpeg-2.1/libavfilter/af_afade.c – Rev 6148