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path: root/libavfilter/af_aphaser.c (plain)
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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
33typedef 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
58static 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
72AVFILTER_DEFINE_CLASS(aphaser);
73
74static 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
86static 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) \
122static 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) \
158static 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
193PHASER_PLANAR(dbl, double)
194PHASER_PLANAR(flt, float)
195PHASER_PLANAR(s16, int16_t)
196PHASER_PLANAR(s32, int32_t)
197
198PHASER(dbl, double)
199PHASER(flt, float)
200PHASER(s16, int16_t)
201PHASER(s32, int32_t)
202
203static 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
241static 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
265static 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
273static const AVFilterPad aphaser_inputs[] = {
274 {
275 .name = "default",
276 .type = AVMEDIA_TYPE_AUDIO,
277 .filter_frame = filter_frame,
278 },
279 { NULL }
280};
281
282static const AVFilterPad aphaser_outputs[] = {
283 {
284 .name = "default",
285 .type = AVMEDIA_TYPE_AUDIO,
286 .config_props = config_output,
287 },
288 { NULL }
289};
290
291AVFilter 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};
302
generated by cgit at 2024-05-17 03:43:34 (GMT)