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
init(AVFilterContext * ctx)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 #define MOD(a, b) (((a) >= (b)) ? (a) - (b) : (a))
87
88 #define PHASER_PLANAR(name, type) \
89 static void phaser_## name ##p(AudioPhaserContext *s, \
90 uint8_t * const *ssrc, uint8_t **ddst, \
91 int nb_samples, int channels) \
92 { \
93 int i, c, delay_pos, modulation_pos; \
94 \
95 av_assert0(channels > 0); \
96 for (c = 0; c < channels; c++) { \
97 type *src = (type *)ssrc[c]; \
98 type *dst = (type *)ddst[c]; \
99 double *buffer = s->delay_buffer + \
100 c * s->delay_buffer_length; \
101 \
102 delay_pos = s->delay_pos; \
103 modulation_pos = s->modulation_pos; \
104 \
105 for (i = 0; i < nb_samples; i++, src++, dst++) { \
106 double v = *src * s->in_gain + buffer[ \
107 MOD(delay_pos + s->modulation_buffer[ \
108 modulation_pos], \
109 s->delay_buffer_length)] * s->decay; \
110 \
111 modulation_pos = MOD(modulation_pos + 1, \
112 s->modulation_buffer_length); \
113 delay_pos = MOD(delay_pos + 1, s->delay_buffer_length); \
114 buffer[delay_pos] = v; \
115 \
116 *dst = v * s->out_gain; \
117 } \
118 } \
119 \
120 s->delay_pos = delay_pos; \
121 s->modulation_pos = modulation_pos; \
122 }
123
124 #define PHASER(name, type) \
125 static void phaser_## name (AudioPhaserContext *s, \
126 uint8_t * const *ssrc, uint8_t **ddst, \
127 int nb_samples, int channels) \
128 { \
129 int i, c, delay_pos, modulation_pos; \
130 type *src = (type *)ssrc[0]; \
131 type *dst = (type *)ddst[0]; \
132 double *buffer = s->delay_buffer; \
133 \
134 delay_pos = s->delay_pos; \
135 modulation_pos = s->modulation_pos; \
136 \
137 for (i = 0; i < nb_samples; i++) { \
138 int pos = MOD(delay_pos + s->modulation_buffer[modulation_pos], \
139 s->delay_buffer_length) * channels; \
140 int npos; \
141 \
142 delay_pos = MOD(delay_pos + 1, s->delay_buffer_length); \
143 npos = delay_pos * channels; \
144 for (c = 0; c < channels; c++, src++, dst++) { \
145 double v = *src * s->in_gain + buffer[pos + c] * s->decay; \
146 \
147 buffer[npos + c] = v; \
148 \
149 *dst = v * s->out_gain; \
150 } \
151 \
152 modulation_pos = MOD(modulation_pos + 1, \
153 s->modulation_buffer_length); \
154 } \
155 \
156 s->delay_pos = delay_pos; \
157 s->modulation_pos = modulation_pos; \
158 }
159
PHASER_PLANAR(dbl,double)160 PHASER_PLANAR(dbl, double)
161 PHASER_PLANAR(flt, float)
162 PHASER_PLANAR(s16, int16_t)
163 PHASER_PLANAR(s32, int32_t)
164
165 PHASER(dbl, double)
166 PHASER(flt, float)
167 PHASER(s16, int16_t)
168 PHASER(s32, int32_t)
169
170 static int config_output(AVFilterLink *outlink)
171 {
172 AudioPhaserContext *s = outlink->src->priv;
173 AVFilterLink *inlink = outlink->src->inputs[0];
174
175 s->delay_buffer_length = s->delay * 0.001 * inlink->sample_rate + 0.5;
176 if (s->delay_buffer_length <= 0) {
177 av_log(outlink->src, AV_LOG_ERROR, "delay is too small\n");
178 return AVERROR(EINVAL);
179 }
180 s->delay_buffer = av_calloc(s->delay_buffer_length, sizeof(*s->delay_buffer) * inlink->ch_layout.nb_channels);
181 s->modulation_buffer_length = inlink->sample_rate / s->speed + 0.5;
182 s->modulation_buffer = av_malloc_array(s->modulation_buffer_length, sizeof(*s->modulation_buffer));
183
184 if (!s->modulation_buffer || !s->delay_buffer)
185 return AVERROR(ENOMEM);
186
187 ff_generate_wave_table(s->type, AV_SAMPLE_FMT_S32,
188 s->modulation_buffer, s->modulation_buffer_length,
189 1., s->delay_buffer_length, M_PI / 2.0);
190
191 s->delay_pos = s->modulation_pos = 0;
192
193 switch (inlink->format) {
194 case AV_SAMPLE_FMT_DBL: s->phaser = phaser_dbl; break;
195 case AV_SAMPLE_FMT_DBLP: s->phaser = phaser_dblp; break;
196 case AV_SAMPLE_FMT_FLT: s->phaser = phaser_flt; break;
197 case AV_SAMPLE_FMT_FLTP: s->phaser = phaser_fltp; break;
198 case AV_SAMPLE_FMT_S16: s->phaser = phaser_s16; break;
199 case AV_SAMPLE_FMT_S16P: s->phaser = phaser_s16p; break;
200 case AV_SAMPLE_FMT_S32: s->phaser = phaser_s32; break;
201 case AV_SAMPLE_FMT_S32P: s->phaser = phaser_s32p; break;
202 default: av_assert0(0);
203 }
204
205 return 0;
206 }
207
filter_frame(AVFilterLink * inlink,AVFrame * inbuf)208 static int filter_frame(AVFilterLink *inlink, AVFrame *inbuf)
209 {
210 AudioPhaserContext *s = inlink->dst->priv;
211 AVFilterLink *outlink = inlink->dst->outputs[0];
212 AVFrame *outbuf;
213
214 if (av_frame_is_writable(inbuf)) {
215 outbuf = inbuf;
216 } else {
217 outbuf = ff_get_audio_buffer(outlink, inbuf->nb_samples);
218 if (!outbuf) {
219 av_frame_free(&inbuf);
220 return AVERROR(ENOMEM);
221 }
222 av_frame_copy_props(outbuf, inbuf);
223 }
224
225 s->phaser(s, inbuf->extended_data, outbuf->extended_data,
226 outbuf->nb_samples, outbuf->ch_layout.nb_channels);
227
228 if (inbuf != outbuf)
229 av_frame_free(&inbuf);
230
231 return ff_filter_frame(outlink, outbuf);
232 }
233
uninit(AVFilterContext * ctx)234 static av_cold void uninit(AVFilterContext *ctx)
235 {
236 AudioPhaserContext *s = ctx->priv;
237
238 av_freep(&s->delay_buffer);
239 av_freep(&s->modulation_buffer);
240 }
241
242 static const AVFilterPad aphaser_inputs[] = {
243 {
244 .name = "default",
245 .type = AVMEDIA_TYPE_AUDIO,
246 .filter_frame = filter_frame,
247 },
248 };
249
250 static const AVFilterPad aphaser_outputs[] = {
251 {
252 .name = "default",
253 .type = AVMEDIA_TYPE_AUDIO,
254 .config_props = config_output,
255 },
256 };
257
258 const AVFilter ff_af_aphaser = {
259 .name = "aphaser",
260 .description = NULL_IF_CONFIG_SMALL("Add a phasing effect to the audio."),
261 .priv_size = sizeof(AudioPhaserContext),
262 .init = init,
263 .uninit = uninit,
264 FILTER_INPUTS(aphaser_inputs),
265 FILTER_OUTPUTS(aphaser_outputs),
266 FILTER_SAMPLEFMTS(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBLP,
267 AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP,
268 AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32P,
269 AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16P),
270 .priv_class = &aphaser_class,
271 };
272