LICENSE.md: add *_rect filters
[ffmpeg.git] / libavfilter / af_aphaser.c
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  * phaser audio filter
24  */
25
26 #include "libavutil/avassert.h"
27 #include "libavutil/opt.h"
28 #include "audio.h"
29 #include "avfilter.h"
30 #include "internal.h"
31 #include "generate_wave_table.h"
32
33 typedef struct AudioPhaserContext {
34     const AVClass *class;
35     double in_gain, out_gain;
36     double delay;
37     double decay;
38     double speed;
39
40     int type;
41
42     int delay_buffer_length;
43     double *delay_buffer;
44
45     int modulation_buffer_length;
46     int32_t *modulation_buffer;
47
48     int delay_pos, modulation_pos;
49
50     void (*phaser)(struct AudioPhaserContext *p,
51                    uint8_t * const *src, uint8_t **dst,
52                    int nb_samples, int channels);
53 } AudioPhaserContext;
54
55 #define OFFSET(x) offsetof(AudioPhaserContext, x)
56 #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
57
58 static const AVOption aphaser_options[] = {
59     { "in_gain",  "set input gain",            OFFSET(in_gain),  AV_OPT_TYPE_DOUBLE, {.dbl=.4},  0,  1,   FLAGS },
60     { "out_gain", "set output gain",           OFFSET(out_gain), AV_OPT_TYPE_DOUBLE, {.dbl=.74}, 0,  1e9, FLAGS },
61     { "delay",    "set delay in milliseconds", OFFSET(delay),    AV_OPT_TYPE_DOUBLE, {.dbl=3.},  0,  5,   FLAGS },
62     { "decay",    "set decay",                 OFFSET(decay),    AV_OPT_TYPE_DOUBLE, {.dbl=.4},  0, .99,  FLAGS },
63     { "speed",    "set modulation speed",      OFFSET(speed),    AV_OPT_TYPE_DOUBLE, {.dbl=.5}, .1,  2,   FLAGS },
64     { "type",     "set modulation type",       OFFSET(type),     AV_OPT_TYPE_INT,    {.i64=WAVE_TRI}, 0, WAVE_NB-1, FLAGS, "type" },
65     { "triangular",  NULL, 0, AV_OPT_TYPE_CONST,  {.i64=WAVE_TRI}, 0, 0, FLAGS, "type" },
66     { "t",           NULL, 0, AV_OPT_TYPE_CONST,  {.i64=WAVE_TRI}, 0, 0, FLAGS, "type" },
67     { "sinusoidal",  NULL, 0, AV_OPT_TYPE_CONST,  {.i64=WAVE_SIN}, 0, 0, FLAGS, "type" },
68     { "s",           NULL, 0, AV_OPT_TYPE_CONST,  {.i64=WAVE_SIN}, 0, 0, FLAGS, "type" },
69     { NULL }
70 };
71
72 AVFILTER_DEFINE_CLASS(aphaser);
73
74 static av_cold int init(AVFilterContext *ctx)
75 {
76     AudioPhaserContext *p = ctx->priv;
77
78     if (p->in_gain > (1 - p->decay * p->decay))
79         av_log(ctx, AV_LOG_WARNING, "in_gain may cause clipping\n");
80     if (p->in_gain / (1 - p->decay) > 1 / p->out_gain)
81         av_log(ctx, AV_LOG_WARNING, "out_gain may cause clipping\n");
82
83     return 0;
84 }
85
86 static int query_formats(AVFilterContext *ctx)
87 {
88     AVFilterFormats *formats;
89     AVFilterChannelLayouts *layouts;
90     static const enum AVSampleFormat sample_fmts[] = {
91         AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBLP,
92         AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP,
93         AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32P,
94         AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16P,
95         AV_SAMPLE_FMT_NONE
96     };
97     int ret;
98
99     layouts = ff_all_channel_layouts();
100     if (!layouts)
101         return AVERROR(ENOMEM);
102     ret = ff_set_common_channel_layouts(ctx, layouts);
103     if (ret < 0)
104         return ret;
105
106     formats = ff_make_format_list(sample_fmts);
107     if (!formats)
108         return AVERROR(ENOMEM);
109     ret = ff_set_common_formats(ctx, formats);
110     if (ret < 0)
111         return ret;
112
113     formats = ff_all_samplerates();
114     if (!formats)
115         return AVERROR(ENOMEM);
116     return ff_set_common_samplerates(ctx, formats);
117 }
118
119 #define MOD(a, b) (((a) >= (b)) ? (a) - (b) : (a))
120
121 #define PHASER_PLANAR(name, type)                                      \
122 static void phaser_## name ##p(AudioPhaserContext *p,                  \
123                                uint8_t * const *src, uint8_t **dst,    \
124                                int nb_samples, int channels)           \
125 {                                                                      \
126     int i, c, delay_pos, modulation_pos;                               \
127                                                                        \
128     av_assert0(channels > 0);                                          \
129     for (c = 0; c < channels; c++) {                                   \
130         type *s = (type *)src[c];                                      \
131         type *d = (type *)dst[c];                                      \
132         double *buffer = p->delay_buffer +                             \
133                          c * p->delay_buffer_length;                   \
134                                                                        \
135         delay_pos      = p->delay_pos;                                 \
136         modulation_pos = p->modulation_pos;                            \
137                                                                        \
138         for (i = 0; i < nb_samples; i++, s++, d++) {                   \
139             double v = *s * p->in_gain + buffer[                       \
140                        MOD(delay_pos + p->modulation_buffer[           \
141                        modulation_pos],                                \
142                        p->delay_buffer_length)] * p->decay;            \
143                                                                        \
144             modulation_pos = MOD(modulation_pos + 1,                   \
145                              p->modulation_buffer_length);             \
146             delay_pos = MOD(delay_pos + 1, p->delay_buffer_length);    \
147             buffer[delay_pos] = v;                                     \
148                                                                        \
149             *d = v * p->out_gain;                                      \
150         }                                                              \
151     }                                                                  \
152                                                                        \
153     p->delay_pos      = delay_pos;                                     \
154     p->modulation_pos = modulation_pos;                                \
155 }
156
157 #define PHASER(name, type)                                              \
158 static void phaser_## name (AudioPhaserContext *p,                      \
159                             uint8_t * const *src, uint8_t **dst,        \
160                             int nb_samples, int channels)               \
161 {                                                                       \
162     int i, c, delay_pos, modulation_pos;                                \
163     type *s = (type *)src[0];                                           \
164     type *d = (type *)dst[0];                                           \
165     double *buffer = p->delay_buffer;                                   \
166                                                                         \
167     delay_pos      = p->delay_pos;                                      \
168     modulation_pos = p->modulation_pos;                                 \
169                                                                         \
170     for (i = 0; i < nb_samples; i++) {                                  \
171         int pos = MOD(delay_pos + p->modulation_buffer[modulation_pos], \
172                    p->delay_buffer_length) * channels;                  \
173         int npos;                                                       \
174                                                                         \
175         delay_pos = MOD(delay_pos + 1, p->delay_buffer_length);         \
176         npos = delay_pos * channels;                                    \
177         for (c = 0; c < channels; c++, s++, d++) {                      \
178             double v = *s * p->in_gain + buffer[pos + c] * p->decay;    \
179                                                                         \
180             buffer[npos + c] = v;                                       \
181                                                                         \
182             *d = v * p->out_gain;                                       \
183         }                                                               \
184                                                                         \
185         modulation_pos = MOD(modulation_pos + 1,                        \
186                          p->modulation_buffer_length);                  \
187     }                                                                   \
188                                                                         \
189     p->delay_pos      = delay_pos;                                      \
190     p->modulation_pos = modulation_pos;                                 \
191 }
192
193 PHASER_PLANAR(dbl, double)
194 PHASER_PLANAR(flt, float)
195 PHASER_PLANAR(s16, int16_t)
196 PHASER_PLANAR(s32, int32_t)
197
198 PHASER(dbl, double)
199 PHASER(flt, float)
200 PHASER(s16, int16_t)
201 PHASER(s32, int32_t)
202
203 static int config_output(AVFilterLink *outlink)
204 {
205     AudioPhaserContext *p = outlink->src->priv;
206     AVFilterLink *inlink = outlink->src->inputs[0];
207
208     p->delay_buffer_length = p->delay * 0.001 * inlink->sample_rate + 0.5;
209     p->delay_buffer = av_calloc(p->delay_buffer_length, sizeof(*p->delay_buffer) * inlink->channels);
210     p->modulation_buffer_length = inlink->sample_rate / p->speed + 0.5;
211     p->modulation_buffer = av_malloc_array(p->modulation_buffer_length, sizeof(*p->modulation_buffer));
212
213     if (!p->modulation_buffer || !p->delay_buffer)
214         return AVERROR(ENOMEM);
215
216     ff_generate_wave_table(p->type, AV_SAMPLE_FMT_S32,
217                            p->modulation_buffer, p->modulation_buffer_length,
218                            1., p->delay_buffer_length, M_PI / 2.0);
219
220     p->delay_pos = p->modulation_pos = 0;
221
222     switch (inlink->format) {
223     case AV_SAMPLE_FMT_DBL:  p->phaser = phaser_dbl;  break;
224     case AV_SAMPLE_FMT_DBLP: p->phaser = phaser_dblp; break;
225     case AV_SAMPLE_FMT_FLT:  p->phaser = phaser_flt;  break;
226     case AV_SAMPLE_FMT_FLTP: p->phaser = phaser_fltp; break;
227     case AV_SAMPLE_FMT_S16:  p->phaser = phaser_s16;  break;
228     case AV_SAMPLE_FMT_S16P: p->phaser = phaser_s16p; break;
229     case AV_SAMPLE_FMT_S32:  p->phaser = phaser_s32;  break;
230     case AV_SAMPLE_FMT_S32P: p->phaser = phaser_s32p; break;
231     default: av_assert0(0);
232     }
233
234     return 0;
235 }
236
237 static int filter_frame(AVFilterLink *inlink, AVFrame *inbuf)
238 {
239     AudioPhaserContext *p = inlink->dst->priv;
240     AVFilterLink *outlink = inlink->dst->outputs[0];
241     AVFrame *outbuf;
242
243     if (av_frame_is_writable(inbuf)) {
244         outbuf = inbuf;
245     } else {
246         outbuf = ff_get_audio_buffer(inlink, inbuf->nb_samples);
247         if (!outbuf)
248             return AVERROR(ENOMEM);
249         av_frame_copy_props(outbuf, inbuf);
250     }
251
252     p->phaser(p, inbuf->extended_data, outbuf->extended_data,
253               outbuf->nb_samples, av_frame_get_channels(outbuf));
254
255     if (inbuf != outbuf)
256         av_frame_free(&inbuf);
257
258     return ff_filter_frame(outlink, outbuf);
259 }
260
261 static av_cold void uninit(AVFilterContext *ctx)
262 {
263     AudioPhaserContext *p = ctx->priv;
264
265     av_freep(&p->delay_buffer);
266     av_freep(&p->modulation_buffer);
267 }
268
269 static const AVFilterPad aphaser_inputs[] = {
270     {
271         .name         = "default",
272         .type         = AVMEDIA_TYPE_AUDIO,
273         .filter_frame = filter_frame,
274     },
275     { NULL }
276 };
277
278 static const AVFilterPad aphaser_outputs[] = {
279     {
280         .name         = "default",
281         .type         = AVMEDIA_TYPE_AUDIO,
282         .config_props = config_output,
283     },
284     { NULL }
285 };
286
287 AVFilter ff_af_aphaser = {
288     .name          = "aphaser",
289     .description   = NULL_IF_CONFIG_SMALL("Add a phasing effect to the audio."),
290     .query_formats = query_formats,
291     .priv_size     = sizeof(AudioPhaserContext),
292     .init          = init,
293     .uninit        = uninit,
294     .inputs        = aphaser_inputs,
295     .outputs       = aphaser_outputs,
296     .priv_class    = &aphaser_class,
297 };