2 * Copyright (c) 2013 Paul B Mahol
4 * This file is part of FFmpeg.
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.
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.
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
26 #include "libavutil/opt.h"
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);
45 enum CurveType { TRI, QSIN, ESIN, HSIN, LOG, PAR, QUA, CUB, SQU, CBR };
47 #define OFFSET(x) offsetof(AudioFadeContext, x)
48 #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
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" },
78 AVFILTER_DEFINE_CLASS(afade);
80 static av_cold int init(AVFilterContext *ctx)
82 AudioFadeContext *afade = ctx->priv;
84 if (INT64_MAX - afade->nb_samples < afade->start_sample)
85 return AVERROR(EINVAL);
90 static int query_formats(AVFilterContext *ctx)
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,
102 layouts = ff_all_channel_layouts();
104 return AVERROR(ENOMEM);
105 ff_set_common_channel_layouts(ctx, layouts);
107 formats = ff_make_format_list(sample_fmts);
109 return AVERROR(ENOMEM);
110 ff_set_common_formats(ctx, formats);
112 formats = ff_all_samplerates();
114 return AVERROR(ENOMEM);
115 ff_set_common_samplerates(ctx, formats);
120 static double fade_gain(int curve, int64_t index, int range)
124 gain = FFMAX(0.0, FFMIN(1.0, 1.0 * index / range));
128 gain = sin(gain * M_PI / 2.0);
131 gain = 1.0 - cos(M_PI / 4.0 * (pow(2.0*gain - 1, 3) + 1));
134 gain = (1.0 - cos(gain * M_PI)) / 2.0;
137 gain = pow(0.1, (1 - gain) * 5.0);
140 gain = (1 - (1 - gain) * (1 - gain));
146 gain = gain * gain * gain;
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) \
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]; \
172 d[i] = s[i] * gain; \
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) \
182 type *d = (type *)dst[0]; \
183 const type *s = (type *)src[0]; \
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; \
193 FADE_PLANAR(dbl, double)
194 FADE_PLANAR(flt, float)
195 FADE_PLANAR(s16, int16_t)
196 FADE_PLANAR(s32, int32_t)
203 static int config_output(AVFilterLink *outlink)
205 AVFilterContext *ctx = outlink->src;
206 AudioFadeContext *afade = ctx->priv;
207 AVFilterLink *inlink = ctx->inputs[0];
209 switch (inlink->format) {
210 case AV_SAMPLE_FMT_DBL: afade->fade_samples = fade_samples_dbl; break;
211 case AV_SAMPLE_FMT_DBLP: afade->fade_samples = fade_samples_dblp; break;
212 case AV_SAMPLE_FMT_FLT: afade->fade_samples = fade_samples_flt; break;
213 case AV_SAMPLE_FMT_FLTP: afade->fade_samples = fade_samples_fltp; break;
214 case AV_SAMPLE_FMT_S16: afade->fade_samples = fade_samples_s16; break;
215 case AV_SAMPLE_FMT_S16P: afade->fade_samples = fade_samples_s16p; break;
216 case AV_SAMPLE_FMT_S32: afade->fade_samples = fade_samples_s32; break;
217 case AV_SAMPLE_FMT_S32P: afade->fade_samples = fade_samples_s32p; break;
221 afade->nb_samples = afade->duration * inlink->sample_rate / AV_TIME_BASE;
222 if (afade->start_time)
223 afade->start_sample = afade->start_time * inlink->sample_rate / AV_TIME_BASE;
228 static int filter_frame(AVFilterLink *inlink, AVFrame *buf)
230 AudioFadeContext *afade = inlink->dst->priv;
231 AVFilterLink *outlink = inlink->dst->outputs[0];
232 int nb_samples = buf->nb_samples;
234 int64_t cur_sample = av_rescale_q(buf->pts, (AVRational){1, outlink->sample_rate}, outlink->time_base);
236 if ((!afade->type && (afade->start_sample + afade->nb_samples < cur_sample)) ||
237 ( afade->type && (cur_sample + afade->nb_samples < afade->start_sample)))
238 return ff_filter_frame(outlink, buf);
240 if (av_frame_is_writable(buf)) {
243 out_buf = ff_get_audio_buffer(inlink, nb_samples);
245 return AVERROR(ENOMEM);
246 av_frame_copy_props(out_buf, buf);
249 if ((!afade->type && (cur_sample + nb_samples < afade->start_sample)) ||
250 ( afade->type && (afade->start_sample + afade->nb_samples < cur_sample))) {
251 av_samples_set_silence(out_buf->extended_data, 0, nb_samples,
252 av_frame_get_channels(out_buf), out_buf->format);
257 start = cur_sample - afade->start_sample;
259 start = afade->start_sample + afade->nb_samples - cur_sample;
261 afade->fade_samples(out_buf->extended_data, buf->extended_data,
262 nb_samples, av_frame_get_channels(buf),
263 afade->type ? -1 : 1, start,
264 afade->nb_samples, afade->curve);
270 return ff_filter_frame(outlink, out_buf);
273 static const AVFilterPad avfilter_af_afade_inputs[] = {
276 .type = AVMEDIA_TYPE_AUDIO,
277 .filter_frame = filter_frame,
282 static const AVFilterPad avfilter_af_afade_outputs[] = {
285 .type = AVMEDIA_TYPE_AUDIO,
286 .config_props = config_output,
291 AVFilter avfilter_af_afade = {
293 .description = NULL_IF_CONFIG_SMALL("Fade in/out input audio."),
294 .query_formats = query_formats,
295 .priv_size = sizeof(AudioFadeContext),
297 .inputs = avfilter_af_afade_inputs,
298 .outputs = avfilter_af_afade_outputs,
299 .priv_class = &afade_class,