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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 | ||
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 | enum WaveType 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 | ||
98 | layouts = ff_all_channel_layouts(); | |
99 | if (!layouts) | |
100 | return AVERROR(ENOMEM); | |
101 | ff_set_common_channel_layouts(ctx, layouts); | |
102 | ||
103 | formats = ff_make_format_list(sample_fmts); | |
104 | if (!formats) | |
105 | return AVERROR(ENOMEM); | |
106 | ff_set_common_formats(ctx, formats); | |
107 | ||
108 | formats = ff_all_samplerates(); | |
109 | if (!formats) | |
110 | return AVERROR(ENOMEM); | |
111 | ff_set_common_samplerates(ctx, formats); | |
112 | ||
113 | return 0; | |
114 | } | |
115 | ||
116 | #define MOD(a, b) (((a) >= (b)) ? (a) - (b) : (a)) | |
117 | ||
118 | #define PHASER_PLANAR(name, type) \ | |
119 | static void phaser_## name ##p(AudioPhaserContext *p, \ | |
120 | uint8_t * const *src, uint8_t **dst, \ | |
121 | int nb_samples, int channels) \ | |
122 | { \ | |
123 | int i, c, delay_pos, modulation_pos; \ | |
124 | \ | |
125 | av_assert0(channels > 0); \ | |
126 | for (c = 0; c < channels; c++) { \ | |
127 | type *s = (type *)src[c]; \ | |
128 | type *d = (type *)dst[c]; \ | |
129 | double *buffer = p->delay_buffer + \ | |
130 | c * p->delay_buffer_length; \ | |
131 | \ | |
132 | delay_pos = p->delay_pos; \ | |
133 | modulation_pos = p->modulation_pos; \ | |
134 | \ | |
135 | for (i = 0; i < nb_samples; i++, s++, d++) { \ | |
136 | double v = *s * p->in_gain + buffer[ \ | |
137 | MOD(delay_pos + p->modulation_buffer[ \ | |
138 | modulation_pos], \ | |
139 | p->delay_buffer_length)] * p->decay; \ | |
140 | \ | |
141 | modulation_pos = MOD(modulation_pos + 1, \ | |
142 | p->modulation_buffer_length); \ | |
143 | delay_pos = MOD(delay_pos + 1, p->delay_buffer_length); \ | |
144 | buffer[delay_pos] = v; \ | |
145 | \ | |
146 | *d = v * p->out_gain; \ | |
147 | } \ | |
148 | } \ | |
149 | \ | |
150 | p->delay_pos = delay_pos; \ | |
151 | p->modulation_pos = modulation_pos; \ | |
152 | } | |
153 | ||
154 | #define PHASER(name, type) \ | |
155 | static void phaser_## name (AudioPhaserContext *p, \ | |
156 | uint8_t * const *src, uint8_t **dst, \ | |
157 | int nb_samples, int channels) \ | |
158 | { \ | |
159 | int i, c, delay_pos, modulation_pos; \ | |
160 | type *s = (type *)src[0]; \ | |
161 | type *d = (type *)dst[0]; \ | |
162 | double *buffer = p->delay_buffer; \ | |
163 | \ | |
164 | delay_pos = p->delay_pos; \ | |
165 | modulation_pos = p->modulation_pos; \ | |
166 | \ | |
167 | for (i = 0; i < nb_samples; i++) { \ | |
168 | int pos = MOD(delay_pos + p->modulation_buffer[modulation_pos], \ | |
169 | p->delay_buffer_length) * channels; \ | |
170 | int npos; \ | |
171 | \ | |
172 | delay_pos = MOD(delay_pos + 1, p->delay_buffer_length); \ | |
173 | npos = delay_pos * channels; \ | |
174 | for (c = 0; c < channels; c++, s++, d++) { \ | |
175 | double v = *s * p->in_gain + buffer[pos + c] * p->decay; \ | |
176 | \ | |
177 | buffer[npos + c] = v; \ | |
178 | \ | |
179 | *d = v * p->out_gain; \ | |
180 | } \ | |
181 | \ | |
182 | modulation_pos = MOD(modulation_pos + 1, \ | |
183 | p->modulation_buffer_length); \ | |
184 | } \ | |
185 | \ | |
186 | p->delay_pos = delay_pos; \ | |
187 | p->modulation_pos = modulation_pos; \ | |
188 | } | |
189 | ||
190 | PHASER_PLANAR(dbl, double) | |
191 | PHASER_PLANAR(flt, float) | |
192 | PHASER_PLANAR(s16, int16_t) | |
193 | PHASER_PLANAR(s32, int32_t) | |
194 | ||
195 | PHASER(dbl, double) | |
196 | PHASER(flt, float) | |
197 | PHASER(s16, int16_t) | |
198 | PHASER(s32, int32_t) | |
199 | ||
200 | static int config_output(AVFilterLink *outlink) | |
201 | { | |
202 | AudioPhaserContext *p = outlink->src->priv; | |
203 | AVFilterLink *inlink = outlink->src->inputs[0]; | |
204 | ||
205 | p->delay_buffer_length = p->delay * 0.001 * inlink->sample_rate + 0.5; | |
206 | p->delay_buffer = av_calloc(p->delay_buffer_length, sizeof(*p->delay_buffer) * inlink->channels); | |
207 | p->modulation_buffer_length = inlink->sample_rate / p->speed + 0.5; | |
208 | p->modulation_buffer = av_malloc_array(p->modulation_buffer_length, sizeof(*p->modulation_buffer)); | |
209 | ||
210 | if (!p->modulation_buffer || !p->delay_buffer) | |
211 | return AVERROR(ENOMEM); | |
212 | ||
213 | ff_generate_wave_table(p->type, AV_SAMPLE_FMT_S32, | |
214 | p->modulation_buffer, p->modulation_buffer_length, | |
215 | 1., p->delay_buffer_length, M_PI / 2.0); | |
216 | ||
217 | p->delay_pos = p->modulation_pos = 0; | |
218 | ||
219 | switch (inlink->format) { | |
220 | case AV_SAMPLE_FMT_DBL: p->phaser = phaser_dbl; break; | |
221 | case AV_SAMPLE_FMT_DBLP: p->phaser = phaser_dblp; break; | |
222 | case AV_SAMPLE_FMT_FLT: p->phaser = phaser_flt; break; | |
223 | case AV_SAMPLE_FMT_FLTP: p->phaser = phaser_fltp; break; | |
224 | case AV_SAMPLE_FMT_S16: p->phaser = phaser_s16; break; | |
225 | case AV_SAMPLE_FMT_S16P: p->phaser = phaser_s16p; break; | |
226 | case AV_SAMPLE_FMT_S32: p->phaser = phaser_s32; break; | |
227 | case AV_SAMPLE_FMT_S32P: p->phaser = phaser_s32p; break; | |
228 | default: av_assert0(0); | |
229 | } | |
230 | ||
231 | return 0; | |
232 | } | |
233 | ||
234 | static int filter_frame(AVFilterLink *inlink, AVFrame *inbuf) | |
235 | { | |
236 | AudioPhaserContext *p = inlink->dst->priv; | |
237 | AVFilterLink *outlink = inlink->dst->outputs[0]; | |
238 | AVFrame *outbuf; | |
239 | ||
240 | if (av_frame_is_writable(inbuf)) { | |
241 | outbuf = inbuf; | |
242 | } else { | |
243 | outbuf = ff_get_audio_buffer(inlink, inbuf->nb_samples); | |
244 | if (!outbuf) | |
245 | return AVERROR(ENOMEM); | |
246 | av_frame_copy_props(outbuf, inbuf); | |
247 | } | |
248 | ||
249 | p->phaser(p, inbuf->extended_data, outbuf->extended_data, | |
250 | outbuf->nb_samples, av_frame_get_channels(outbuf)); | |
251 | ||
252 | if (inbuf != outbuf) | |
253 | av_frame_free(&inbuf); | |
254 | ||
255 | return ff_filter_frame(outlink, outbuf); | |
256 | } | |
257 | ||
258 | static av_cold void uninit(AVFilterContext *ctx) | |
259 | { | |
260 | AudioPhaserContext *p = ctx->priv; | |
261 | ||
262 | av_freep(&p->delay_buffer); | |
263 | av_freep(&p->modulation_buffer); | |
264 | } | |
265 | ||
266 | static const AVFilterPad aphaser_inputs[] = { | |
267 | { | |
268 | .name = "default", | |
269 | .type = AVMEDIA_TYPE_AUDIO, | |
270 | .filter_frame = filter_frame, | |
271 | }, | |
272 | { NULL } | |
273 | }; | |
274 | ||
275 | static const AVFilterPad aphaser_outputs[] = { | |
276 | { | |
277 | .name = "default", | |
278 | .type = AVMEDIA_TYPE_AUDIO, | |
279 | .config_props = config_output, | |
280 | }, | |
281 | { NULL } | |
282 | }; | |
283 | ||
284 | AVFilter ff_af_aphaser = { | |
285 | .name = "aphaser", | |
286 | .description = NULL_IF_CONFIG_SMALL("Add a phasing effect to the audio."), | |
287 | .query_formats = query_formats, | |
288 | .priv_size = sizeof(AudioPhaserContext), | |
289 | .init = init, | |
290 | .uninit = uninit, | |
291 | .inputs = aphaser_inputs, | |
292 | .outputs = aphaser_outputs, | |
293 | .priv_class = &aphaser_class, | |
294 | }; |