avformat/utils: add missing brackets around arguments in av_realloc() call
[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 *s,
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 *s = ctx->priv;
77
78     if (s->in_gain > (1 - s->decay * s->decay))
79         av_log(ctx, AV_LOG_WARNING, "in_gain may cause clipping\n");
80     if (s->in_gain / (1 - s->decay) > 1 / s->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_counts();
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 *s,                  \
123                                uint8_t * const *ssrc, uint8_t **ddst,  \
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 *src = (type *)ssrc[c];                                   \
131         type *dst = (type *)ddst[c];                                   \
132         double *buffer = s->delay_buffer +                             \
133                          c * s->delay_buffer_length;                   \
134                                                                        \
135         delay_pos      = s->delay_pos;                                 \
136         modulation_pos = s->modulation_pos;                            \
137                                                                        \
138         for (i = 0; i < nb_samples; i++, src++, dst++) {               \
139             double v = *src * s->in_gain + buffer[                     \
140                        MOD(delay_pos + s->modulation_buffer[           \
141                        modulation_pos],                                \
142                        s->delay_buffer_length)] * s->decay;            \
143                                                                        \
144             modulation_pos = MOD(modulation_pos + 1,                   \
145                              s->modulation_buffer_length);             \
146             delay_pos = MOD(delay_pos + 1, s->delay_buffer_length);    \
147             buffer[delay_pos] = v;                                     \
148                                                                        \
149             *dst = v * s->out_gain;                                    \
150         }                                                              \
151     }                                                                  \
152                                                                        \
153     s->delay_pos      = delay_pos;                                     \
154     s->modulation_pos = modulation_pos;                                \
155 }
156
157 #define PHASER(name, type)                                              \
158 static void phaser_## name (AudioPhaserContext *s,                      \
159                             uint8_t * const *ssrc, uint8_t **ddst,      \
160                             int nb_samples, int channels)               \
161 {                                                                       \
162     int i, c, delay_pos, modulation_pos;                                \
163     type *src = (type *)ssrc[0];                                        \
164     type *dst = (type *)ddst[0];                                        \
165     double *buffer = s->delay_buffer;                                   \
166                                                                         \
167     delay_pos      = s->delay_pos;                                      \
168     modulation_pos = s->modulation_pos;                                 \
169                                                                         \
170     for (i = 0; i < nb_samples; i++) {                                  \
171         int pos = MOD(delay_pos + s->modulation_buffer[modulation_pos], \
172                       s->delay_buffer_length) * channels;               \
173         int npos;                                                       \
174                                                                         \
175         delay_pos = MOD(delay_pos + 1, s->delay_buffer_length);         \
176         npos = delay_pos * channels;                                    \
177         for (c = 0; c < channels; c++, src++, dst++) {                  \
178             double v = *src * s->in_gain + buffer[pos + c] * s->decay;  \
179                                                                         \
180             buffer[npos + c] = v;                                       \
181                                                                         \
182             *dst = v * s->out_gain;                                     \
183         }                                                               \
184                                                                         \
185         modulation_pos = MOD(modulation_pos + 1,                        \
186                          s->modulation_buffer_length);                  \
187     }                                                                   \
188                                                                         \
189     s->delay_pos      = delay_pos;                                      \
190     s->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 *s = outlink->src->priv;
206     AVFilterLink *inlink = outlink->src->inputs[0];
207
208     s->delay_buffer_length = s->delay * 0.001 * inlink->sample_rate + 0.5;
209     if (s->delay_buffer_length <= 0) {
210         av_log(outlink->src, AV_LOG_ERROR, "delay is too small\n");
211         return AVERROR(EINVAL);
212     }
213     s->delay_buffer = av_calloc(s->delay_buffer_length, sizeof(*s->delay_buffer) * inlink->channels);
214     s->modulation_buffer_length = inlink->sample_rate / s->speed + 0.5;
215     s->modulation_buffer = av_malloc_array(s->modulation_buffer_length, sizeof(*s->modulation_buffer));
216
217     if (!s->modulation_buffer || !s->delay_buffer)
218         return AVERROR(ENOMEM);
219
220     ff_generate_wave_table(s->type, AV_SAMPLE_FMT_S32,
221                            s->modulation_buffer, s->modulation_buffer_length,
222                            1., s->delay_buffer_length, M_PI / 2.0);
223
224     s->delay_pos = s->modulation_pos = 0;
225
226     switch (inlink->format) {
227     case AV_SAMPLE_FMT_DBL:  s->phaser = phaser_dbl;  break;
228     case AV_SAMPLE_FMT_DBLP: s->phaser = phaser_dblp; break;
229     case AV_SAMPLE_FMT_FLT:  s->phaser = phaser_flt;  break;
230     case AV_SAMPLE_FMT_FLTP: s->phaser = phaser_fltp; break;
231     case AV_SAMPLE_FMT_S16:  s->phaser = phaser_s16;  break;
232     case AV_SAMPLE_FMT_S16P: s->phaser = phaser_s16p; break;
233     case AV_SAMPLE_FMT_S32:  s->phaser = phaser_s32;  break;
234     case AV_SAMPLE_FMT_S32P: s->phaser = phaser_s32p; break;
235     default: av_assert0(0);
236     }
237
238     return 0;
239 }
240
241 static int filter_frame(AVFilterLink *inlink, AVFrame *inbuf)
242 {
243     AudioPhaserContext *s = inlink->dst->priv;
244     AVFilterLink *outlink = inlink->dst->outputs[0];
245     AVFrame *outbuf;
246
247     if (av_frame_is_writable(inbuf)) {
248         outbuf = inbuf;
249     } else {
250         outbuf = ff_get_audio_buffer(inlink, inbuf->nb_samples);
251         if (!outbuf)
252             return AVERROR(ENOMEM);
253         av_frame_copy_props(outbuf, inbuf);
254     }
255
256     s->phaser(s, inbuf->extended_data, outbuf->extended_data,
257               outbuf->nb_samples, av_frame_get_channels(outbuf));
258
259     if (inbuf != outbuf)
260         av_frame_free(&inbuf);
261
262     return ff_filter_frame(outlink, outbuf);
263 }
264
265 static av_cold void uninit(AVFilterContext *ctx)
266 {
267     AudioPhaserContext *s = ctx->priv;
268
269     av_freep(&s->delay_buffer);
270     av_freep(&s->modulation_buffer);
271 }
272
273 static const AVFilterPad aphaser_inputs[] = {
274     {
275         .name         = "default",
276         .type         = AVMEDIA_TYPE_AUDIO,
277         .filter_frame = filter_frame,
278     },
279     { NULL }
280 };
281
282 static const AVFilterPad aphaser_outputs[] = {
283     {
284         .name         = "default",
285         .type         = AVMEDIA_TYPE_AUDIO,
286         .config_props = config_output,
287     },
288     { NULL }
289 };
290
291 AVFilter ff_af_aphaser = {
292     .name          = "aphaser",
293     .description   = NULL_IF_CONFIG_SMALL("Add a phasing effect to the audio."),
294     .query_formats = query_formats,
295     .priv_size     = sizeof(AudioPhaserContext),
296     .init          = init,
297     .uninit        = uninit,
298     .inputs        = aphaser_inputs,
299     .outputs       = aphaser_outputs,
300     .priv_class    = &aphaser_class,
301 };