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afade filter

Paul B Mahol committed 13 years ago

Commit 6ea8a831 parent 5f61e096 files · +389 −2Browse files

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6 files
  1. Changelog +1 −0
  2. doc/filters.texi +77 −0
  3. libavfilter/Makefile +1 −0
  4. libavfilter/af_afade.c +307 −0
  5. libavfilter/allfilters.c +1 −0
  6. libavfilter/version.h +2 −2
Changelog+1−0
@@ -8,6 +8,7 @@ version <next>: · 7 unmodified lines
8- Chained Ogg support8- Chained Ogg support
9- Theora Midstream reconfiguration support9- Theora Midstream reconfiguration support
10- EVRC decoder10- EVRC decoder
11- audio fade filter
11 12
12 13
13version 1.1:14version 1.1:
doc/filters.texi+77−0
@@ -282,6 +282,83 @@ aconvert=u8:auto · 281 unmodified lines
282@end example282@end example
283@end itemize283@end itemize
284 284
285@section afade
286
287Apply fade-in/out effect to input audio.
288
289The filter accepts parameters as a list of @var{key}=@var{value}
290pairs, separated by ":".
291
292A description of the accepted parameters follows.
293
294@table @option
295@item type, t
296Specify the effect type, can be either @code{in} for fade-in, or
297@code{out} for a fade-out effect. Default is @code{in}.
298
299@item start_sample, ss
300Specify the number of the start sample for starting to apply the fade
301effect. Default is 0.
302
303@item nb_samples, ns
304Specify the number of samples for which the fade effect has to last. At
305the end of the fade-in effect the output audio will have the same
306volume as the input audio, at the end of the fade-out transition
307the output audio will be silence. Default is 44100.
308
309@item start_time, st
310Specify time in seconds for starting to apply the fade
311effect. Default is 0.
312If set this option is used instead of @var{start_sample} one.
313
314@item duration, d
315Specify the number of seconds for which the fade effect has to last. At
316the end of the fade-in effect the output audio will have the same
317volume as the input audio, at the end of the fade-out transition
318the output audio will be silence. Default is 0.
319If set this option is used instead of @var{nb_samples} one.
320
321@item curve
322Set cuve for fade transition.
323@table @option
324@item @var{triangular, linear slope (default)}
325@code{tri}
326@item @var{quarter of sine wave}
327@code{qsin}
328@item @var{half of sine wave}
329@code{esin}
330@item @var{exponential sine wave}
331@code{hsin}
332@item @var{logarithmic}
333@code{log}
334@item @var{inverted parabola}
335@code{par}
336@item @var{quadratic}
337@code{qua}
338@item @var{cubic}
339@code{cub}
340@item @var{square root}
341@code{squ}
342@item @var{cubic root}
343@code{cbr}
344@end table
345@end table
346
347@subsection Examples
348@itemize
349@item
350Fade in first 15 seconds of audio:
351@example
352afade=t=in:ss=0:d=15
353@end example
354
355@item
356Fade out last 25 seconds of a 900 seconds audio:
357@example
358afade=t=out:ss=875:d=25
359@end example
360@end itemize
361
285@section aformat362@section aformat
286 363
287Set output format constraints for the input audio. The framework will364Set output format constraints for the input audio. The framework will
libavfilter/Makefile+1−0
@@ -51,6 +51,7 @@ OBJS-$(CONFIG_AVFORMAT) += lavfutils.o · 50 unmodified lines
51OBJS-$(CONFIG_SWSCALE) += lswsutils.o51OBJS-$(CONFIG_SWSCALE) += lswsutils.o
52 52
53OBJS-$(CONFIG_ACONVERT_FILTER) += af_aconvert.o53OBJS-$(CONFIG_ACONVERT_FILTER) += af_aconvert.o
54OBJS-$(CONFIG_AFADE_FILTER) += af_afade.o
54OBJS-$(CONFIG_AFORMAT_FILTER) += af_aformat.o55OBJS-$(CONFIG_AFORMAT_FILTER) += af_aformat.o
55OBJS-$(CONFIG_AMERGE_FILTER) += af_amerge.o56OBJS-$(CONFIG_AMERGE_FILTER) += af_amerge.o
56OBJS-$(CONFIG_AMIX_FILTER) += af_amix.o57OBJS-$(CONFIG_AMIX_FILTER) += af_amix.o
libavfilter/af_afade.c+307−0
@@ -0,0 +1,307 @@
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
31typedef struct {
32 const AVClass *class;
33 int type;
34 int curve;
35 int nb_samples;
36 int64_t start_sample;
37 double duration;
38 double 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
45enum 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
50static 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", NULL, 0, AV_OPT_TYPE_CONST, {.i64 = 0 }, 0, 0, FLAGS, "type" },
54 { "out", NULL, 0, AV_OPT_TYPE_CONST, {.i64 = 1 }, 0, 0, FLAGS, "type" },
55 { "start_sample", "set expression of sample to start fading", OFFSET(start_sample), AV_OPT_TYPE_INT64, {.i64 = 0 }, 0, INT64_MAX, FLAGS },
56 { "ss", "set expression of sample to start fading", OFFSET(start_sample), AV_OPT_TYPE_INT64, {.i64 = 0 }, 0, INT64_MAX, FLAGS },
57 { "nb_samples", "set expression for fade duration in samples", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64 = 44100}, 1, INT32_MAX, FLAGS },
58 { "ns", "set expression for fade duration in samples", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64 = 44100}, 1, INT32_MAX, FLAGS },
59 { "start_time", "set expression of second to start fading", OFFSET(start_time), AV_OPT_TYPE_DOUBLE, {.dbl = 0. }, 0, 7*24*60*60,FLAGS },
60 { "st", "set expression of second to start fading", OFFSET(start_time), AV_OPT_TYPE_DOUBLE, {.dbl = 0. }, 0, 7*24*60*60,FLAGS },
61 { "duration", "set expression for fade duration in seconds", OFFSET(duration), AV_OPT_TYPE_DOUBLE, {.dbl = 0. }, 0, 24*60*60, FLAGS },
62 { "d", "set expression for fade duration in seconds", OFFSET(duration), AV_OPT_TYPE_DOUBLE, {.dbl = 0. }, 0, 24*60*60, FLAGS },
63 { "curve", "set expression for fade curve", OFFSET(curve), AV_OPT_TYPE_INT, {.i64 = TRI }, TRI, CBR, FLAGS, "curve" },
64 { "c", "set expression for fade curve", 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
78AVFILTER_DEFINE_CLASS(afade);
79
80static av_cold int init(AVFilterContext *ctx, const char *args)
81{
82 AudioFadeContext *afade = ctx->priv;
83 int ret;
84
85 afade->class = &afade_class;
86 av_opt_set_defaults(afade);
87
88 if ((ret = av_set_options_string(afade, args, "=", ":")) < 0)
89 return ret;
90
91 if (INT64_MAX - afade->nb_samples < afade->start_sample)
92 return AVERROR(EINVAL);
93
94 return 0;
95}
96
97static int query_formats(AVFilterContext *ctx)
98{
99 AVFilterFormats *formats;
100 AVFilterChannelLayouts *layouts;
101 static const enum AVSampleFormat sample_fmts[] = {
102 AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16P,
103 AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32P,
104 AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP,
105 AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBLP,
106 AV_SAMPLE_FMT_NONE
107 };
108
109 layouts = ff_all_channel_layouts();
110 if (!layouts)
111 return AVERROR(ENOMEM);
112 ff_set_common_channel_layouts(ctx, layouts);
113
114 formats = ff_make_format_list(sample_fmts);
115 if (!formats)
116 return AVERROR(ENOMEM);
117 ff_set_common_formats(ctx, formats);
118
119 formats = ff_all_samplerates();
120 if (!formats)
121 return AVERROR(ENOMEM);
122 ff_set_common_samplerates(ctx, formats);
123
124 return 0;
125}
126
127static double fade_gain(int curve, int64_t index, int range)
128{
129 double gain;
130
131 gain = FFMAX(0.0, FFMIN(1.0, 1.0 * index / range));
132
133 switch (curve) {
134 case QSIN:
135 gain = sin(gain * M_PI / 2.0);
136 break;
137 case ESIN:
138 gain = 1.0 - cos(M_PI / 4.0 * (pow(2.0*gain - 1, 3) + 1));
139 break;
140 case HSIN:
141 gain = (1.0 - cos(gain * M_PI)) / 2.0;
142 break;
143 case LOG:
144 gain = pow(0.1, (1 - gain) * 5.0);
145 break;
146 case PAR:
147 gain = (1 - (1 - gain) * (1 - gain));
148 break;
149 case QUA:
150 gain *= gain;
151 break;
152 case CUB:
153 gain = gain * gain * gain;
154 break;
155 case SQU:
156 gain = sqrt(gain);
157 break;
158 case CBR:
159 gain = cbrt(gain);
160 break;
161 }
162
163 return gain;
164}
165
166#define FADE_PLANAR(name, type) \
167static void fade_samples_## name ##p(uint8_t **dst, uint8_t * const *src, \
168 int nb_samples, int channels, int dir, \
169 int64_t start, int range, int curve) \
170{ \
171 int i, c; \
172 \
173 for (i = 0; i < nb_samples; i++) { \
174 double gain = fade_gain(curve, start + i * dir, range); \
175 for (c = 0; c < channels; c++) { \
176 type *d = (type *)dst[c]; \
177 const type *s = (type *)src[c]; \
178 \
179 d[i] = s[i] * gain; \
180 } \
181 } \
182}
183
184#define FADE(name, type) \
185static void fade_samples_## name (uint8_t **dst, uint8_t * const *src, \
186 int nb_samples, int channels, int dir, \
187 int64_t start, int range, int curve) \
188{ \
189 type *d = (type *)dst[0]; \
190 const type *s = (type *)src[0]; \
191 int i, c, k = 0; \
192 \
193 for (i = 0; i < nb_samples; i++) { \
194 double gain = fade_gain(curve, start + i * dir, range); \
195 for (c = 0; c < channels; c++, k++) \
196 d[k] = s[k] * gain; \
197 } \
198}
199
200FADE_PLANAR(dbl, double)
201FADE_PLANAR(flt, float)
202FADE_PLANAR(s16, int16_t)
203FADE_PLANAR(s32, int32_t)
204
205FADE(dbl, double)
206FADE(flt, float)
207FADE(s16, int16_t)
208FADE(s32, int32_t)
209
210static int config_output(AVFilterLink *outlink)
211{
212 AVFilterContext *ctx = outlink->src;
213 AudioFadeContext *afade = ctx->priv;
214 AVFilterLink *inlink = ctx->inputs[0];
215
216 switch (inlink->format) {
217 case AV_SAMPLE_FMT_DBL: afade->fade_samples = fade_samples_dbl; break;
218 case AV_SAMPLE_FMT_DBLP: afade->fade_samples = fade_samples_dblp; break;
219 case AV_SAMPLE_FMT_FLT: afade->fade_samples = fade_samples_flt; break;
220 case AV_SAMPLE_FMT_FLTP: afade->fade_samples = fade_samples_fltp; break;
221 case AV_SAMPLE_FMT_S16: afade->fade_samples = fade_samples_s16; break;
222 case AV_SAMPLE_FMT_S16P: afade->fade_samples = fade_samples_s16p; break;
223 case AV_SAMPLE_FMT_S32: afade->fade_samples = fade_samples_s32; break;
224 case AV_SAMPLE_FMT_S32P: afade->fade_samples = fade_samples_s32p; break;
225 }
226
227 if (afade->duration)
228 afade->nb_samples = afade->duration * inlink->sample_rate;
229 if (afade->start_time)
230 afade->start_sample = afade->start_time * inlink->sample_rate;
231
232 return 0;
233}
234
235static int filter_frame(AVFilterLink *inlink, AVFilterBufferRef *buf)
236{
237 AudioFadeContext *afade = inlink->dst->priv;
238 AVFilterLink *outlink = inlink->dst->outputs[0];
239 int nb_samples = buf->audio->nb_samples;
240 AVFilterBufferRef *out_buf;
241 int64_t cur_sample = av_rescale_q(buf->pts, (AVRational){1, outlink->sample_rate}, outlink->time_base);
242
243 if ((!afade->type && (afade->start_sample + afade->nb_samples < cur_sample)) ||
244 ( afade->type && (cur_sample + afade->nb_samples < afade->start_sample)))
245 return ff_filter_frame(outlink, buf);
246
247 if (buf->perms & AV_PERM_WRITE) {
248 out_buf = buf;
249 } else {
250 out_buf = ff_get_audio_buffer(inlink, AV_PERM_WRITE, nb_samples);
251 if (!out_buf)
252 return AVERROR(ENOMEM);
253 out_buf->pts = buf->pts;
254 }
255
256 if ((!afade->type && (cur_sample + nb_samples < afade->start_sample)) ||
257 ( afade->type && (afade->start_sample + afade->nb_samples < cur_sample))) {
258 av_samples_set_silence(out_buf->extended_data, 0, nb_samples,
259 out_buf->audio->channels, out_buf->format);
260 } else {
261 int64_t start;
262
263 if (!afade->type)
264 start = cur_sample - afade->start_sample;
265 else
266 start = afade->start_sample + afade->nb_samples - cur_sample;
267
268 afade->fade_samples(out_buf->extended_data, buf->extended_data,
269 nb_samples, buf->audio->channels,
270 afade->type ? -1 : 1, start,
271 afade->nb_samples, afade->curve);
272 }
273
274 if (buf != out_buf)
275 avfilter_unref_buffer(buf);
276
277 return ff_filter_frame(outlink, out_buf);
278}
279
280static const AVFilterPad avfilter_af_afade_inputs[] = {
281 {
282 .name = "default",
283 .type = AVMEDIA_TYPE_AUDIO,
284 .filter_frame = filter_frame,
285 },
286 { NULL }
287};
288
289static const AVFilterPad avfilter_af_afade_outputs[] = {
290 {
291 .name = "default",
292 .type = AVMEDIA_TYPE_AUDIO,
293 .config_props = config_output,
294 },
295 { NULL }
296};
297
298AVFilter avfilter_af_afade = {
299 .name = "afade",
300 .description = NULL_IF_CONFIG_SMALL("Fade in/out input audio."),
301 .query_formats = query_formats,
302 .priv_size = sizeof(AudioFadeContext),
303 .init = init,
304 .inputs = avfilter_af_afade_inputs,
305 .outputs = avfilter_af_afade_outputs,
306 .priv_class = &afade_class,
307};
libavfilter/allfilters.c+1−0
@@ -45,6 +45,7 @@ void avfilter_register_all(void) · 44 unmodified lines
45 initialized = 1;45 initialized = 1;
46 46
47 REGISTER_FILTER(ACONVERT, aconvert, af);47 REGISTER_FILTER(ACONVERT, aconvert, af);
48 REGISTER_FILTER(AFADE, afade, af);
48 REGISTER_FILTER(AFORMAT, aformat, af);49 REGISTER_FILTER(AFORMAT, aformat, af);
49 REGISTER_FILTER(AMERGE, amerge, af);50 REGISTER_FILTER(AMERGE, amerge, af);
50 REGISTER_FILTER(AMIX, amix, af);51 REGISTER_FILTER(AMIX, amix, af);
libavfilter/version.h+2−2
@@ -29,8 +29,8 @@ · 28 unmodified lines
29#include "libavutil/avutil.h"29#include "libavutil/avutil.h"
30 30
31#define LIBAVFILTER_VERSION_MAJOR 331#define LIBAVFILTER_VERSION_MAJOR 3
32#define LIBAVFILTER_VERSION_MINOR 3232#define LIBAVFILTER_VERSION_MINOR 33
33#define LIBAVFILTER_VERSION_MICRO 10133#define LIBAVFILTER_VERSION_MICRO 100
34 34
35#define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \35#define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \
36 LIBAVFILTER_VERSION_MINOR, \36 LIBAVFILTER_VERSION_MINOR, \

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