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path: root/libavfilter/af_chorus.c (plain)
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1/*
2 * Copyright (c) 1998 Juergen Mueller And Sundry Contributors
3 * This source code is freely redistributable and may be used for
4 * any purpose. This copyright notice must be maintained.
5 * Juergen Mueller And Sundry Contributors are not responsible for
6 * the consequences of using this software.
7 *
8 * Copyright (c) 2015 Paul B Mahol
9 *
10 * This file is part of FFmpeg.
11 *
12 * FFmpeg is free software; you can redistribute it and/or
13 * modify it under the terms of the GNU Lesser General Public
14 * License as published by the Free Software Foundation; either
15 * version 2.1 of the License, or (at your option) any later version.
16 *
17 * FFmpeg is distributed in the hope that it will be useful,
18 * but WITHOUT ANY WARRANTY; without even the implied warranty of
19 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
20 * Lesser General Public License for more details.
21 *
22 * You should have received a copy of the GNU Lesser General Public
23 * License along with FFmpeg; if not, write to the Free Software
24 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
25 */
26
27/**
28 * @file
29 * chorus audio filter
30 */
31
32#include "libavutil/avstring.h"
33#include "libavutil/opt.h"
34#include "audio.h"
35#include "avfilter.h"
36#include "internal.h"
37#include "generate_wave_table.h"
38
39typedef struct ChorusContext {
40 const AVClass *class;
41 float in_gain, out_gain;
42 char *delays_str;
43 char *decays_str;
44 char *speeds_str;
45 char *depths_str;
46 float *delays;
47 float *decays;
48 float *speeds;
49 float *depths;
50 uint8_t **chorusbuf;
51 int **phase;
52 int *length;
53 int32_t **lookup_table;
54 int *counter;
55 int num_chorus;
56 int max_samples;
57 int channels;
58 int modulation;
59 int fade_out;
60 int64_t next_pts;
61} ChorusContext;
62
63#define OFFSET(x) offsetof(ChorusContext, x)
64#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
65
66static const AVOption chorus_options[] = {
67 { "in_gain", "set input gain", OFFSET(in_gain), AV_OPT_TYPE_FLOAT, {.dbl=.4}, 0, 1, A },
68 { "out_gain", "set output gain", OFFSET(out_gain), AV_OPT_TYPE_FLOAT, {.dbl=.4}, 0, 1, A },
69 { "delays", "set delays", OFFSET(delays_str), AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, A },
70 { "decays", "set decays", OFFSET(decays_str), AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, A },
71 { "speeds", "set speeds", OFFSET(speeds_str), AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, A },
72 { "depths", "set depths", OFFSET(depths_str), AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, A },
73 { NULL }
74};
75
76AVFILTER_DEFINE_CLASS(chorus);
77
78static void count_items(char *item_str, int *nb_items)
79{
80 char *p;
81
82 *nb_items = 1;
83 for (p = item_str; *p; p++) {
84 if (*p == '|')
85 (*nb_items)++;
86 }
87
88}
89
90static void fill_items(char *item_str, int *nb_items, float *items)
91{
92 char *p, *saveptr = NULL;
93 int i, new_nb_items = 0;
94
95 p = item_str;
96 for (i = 0; i < *nb_items; i++) {
97 char *tstr = av_strtok(p, "|", &saveptr);
98 p = NULL;
99 if (tstr)
100 new_nb_items += sscanf(tstr, "%f", &items[new_nb_items]) == 1;
101 }
102
103 *nb_items = new_nb_items;
104}
105
106static av_cold int init(AVFilterContext *ctx)
107{
108 ChorusContext *s = ctx->priv;
109 int nb_delays, nb_decays, nb_speeds, nb_depths;
110
111 if (!s->delays_str || !s->decays_str || !s->speeds_str || !s->depths_str) {
112 av_log(ctx, AV_LOG_ERROR, "Both delays & decays & speeds & depths must be set.\n");
113 return AVERROR(EINVAL);
114 }
115
116 count_items(s->delays_str, &nb_delays);
117 count_items(s->decays_str, &nb_decays);
118 count_items(s->speeds_str, &nb_speeds);
119 count_items(s->depths_str, &nb_depths);
120
121 s->delays = av_realloc_f(s->delays, nb_delays, sizeof(*s->delays));
122 s->decays = av_realloc_f(s->decays, nb_decays, sizeof(*s->decays));
123 s->speeds = av_realloc_f(s->speeds, nb_speeds, sizeof(*s->speeds));
124 s->depths = av_realloc_f(s->depths, nb_depths, sizeof(*s->depths));
125
126 if (!s->delays || !s->decays || !s->speeds || !s->depths)
127 return AVERROR(ENOMEM);
128
129 fill_items(s->delays_str, &nb_delays, s->delays);
130 fill_items(s->decays_str, &nb_decays, s->decays);
131 fill_items(s->speeds_str, &nb_speeds, s->speeds);
132 fill_items(s->depths_str, &nb_depths, s->depths);
133
134 if (nb_delays != nb_decays && nb_delays != nb_speeds && nb_delays != nb_depths) {
135 av_log(ctx, AV_LOG_ERROR, "Number of delays & decays & speeds & depths given must be same.\n");
136 return AVERROR(EINVAL);
137 }
138
139 s->num_chorus = nb_delays;
140
141 if (s->num_chorus < 1) {
142 av_log(ctx, AV_LOG_ERROR, "At least one delay & decay & speed & depth must be set.\n");
143 return AVERROR(EINVAL);
144 }
145
146 s->length = av_calloc(s->num_chorus, sizeof(*s->length));
147 s->lookup_table = av_calloc(s->num_chorus, sizeof(*s->lookup_table));
148
149 if (!s->length || !s->lookup_table)
150 return AVERROR(ENOMEM);
151
152 s->next_pts = AV_NOPTS_VALUE;
153
154 return 0;
155}
156
157static int query_formats(AVFilterContext *ctx)
158{
159 AVFilterFormats *formats;
160 AVFilterChannelLayouts *layouts;
161 static const enum AVSampleFormat sample_fmts[] = {
162 AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE
163 };
164 int ret;
165
166 layouts = ff_all_channel_counts();
167 if (!layouts)
168 return AVERROR(ENOMEM);
169 ret = ff_set_common_channel_layouts(ctx, layouts);
170 if (ret < 0)
171 return ret;
172
173 formats = ff_make_format_list(sample_fmts);
174 if (!formats)
175 return AVERROR(ENOMEM);
176 ret = ff_set_common_formats(ctx, formats);
177 if (ret < 0)
178 return ret;
179
180 formats = ff_all_samplerates();
181 if (!formats)
182 return AVERROR(ENOMEM);
183 return ff_set_common_samplerates(ctx, formats);
184}
185
186static int config_output(AVFilterLink *outlink)
187{
188 AVFilterContext *ctx = outlink->src;
189 ChorusContext *s = ctx->priv;
190 float sum_in_volume = 1.0;
191 int n;
192
193 s->channels = outlink->channels;
194
195 for (n = 0; n < s->num_chorus; n++) {
196 int samples = (int) ((s->delays[n] + s->depths[n]) * outlink->sample_rate / 1000.0);
197 int depth_samples = (int) (s->depths[n] * outlink->sample_rate / 1000.0);
198
199 s->length[n] = outlink->sample_rate / s->speeds[n];
200
201 s->lookup_table[n] = av_malloc(sizeof(int32_t) * s->length[n]);
202 if (!s->lookup_table[n])
203 return AVERROR(ENOMEM);
204
205 ff_generate_wave_table(WAVE_SIN, AV_SAMPLE_FMT_S32, s->lookup_table[n],
206 s->length[n], 0., depth_samples, 0);
207 s->max_samples = FFMAX(s->max_samples, samples);
208 }
209
210 for (n = 0; n < s->num_chorus; n++)
211 sum_in_volume += s->decays[n];
212
213 if (s->in_gain * (sum_in_volume) > 1.0 / s->out_gain)
214 av_log(ctx, AV_LOG_WARNING, "output gain can cause saturation or clipping of output\n");
215
216 s->counter = av_calloc(outlink->channels, sizeof(*s->counter));
217 if (!s->counter)
218 return AVERROR(ENOMEM);
219
220 s->phase = av_calloc(outlink->channels, sizeof(*s->phase));
221 if (!s->phase)
222 return AVERROR(ENOMEM);
223
224 for (n = 0; n < outlink->channels; n++) {
225 s->phase[n] = av_calloc(s->num_chorus, sizeof(int));
226 if (!s->phase[n])
227 return AVERROR(ENOMEM);
228 }
229
230 s->fade_out = s->max_samples;
231
232 return av_samples_alloc_array_and_samples(&s->chorusbuf, NULL,
233 outlink->channels,
234 s->max_samples,
235 outlink->format, 0);
236}
237
238#define MOD(a, b) (((a) >= (b)) ? (a) - (b) : (a))
239
240static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
241{
242 AVFilterContext *ctx = inlink->dst;
243 ChorusContext *s = ctx->priv;
244 AVFrame *out_frame;
245 int c, i, n;
246
247 if (av_frame_is_writable(frame)) {
248 out_frame = frame;
249 } else {
250 out_frame = ff_get_audio_buffer(inlink, frame->nb_samples);
251 if (!out_frame) {
252 av_frame_free(&frame);
253 return AVERROR(ENOMEM);
254 }
255 av_frame_copy_props(out_frame, frame);
256 }
257
258 for (c = 0; c < inlink->channels; c++) {
259 const float *src = (const float *)frame->extended_data[c];
260 float *dst = (float *)out_frame->extended_data[c];
261 float *chorusbuf = (float *)s->chorusbuf[c];
262 int *phase = s->phase[c];
263
264 for (i = 0; i < frame->nb_samples; i++) {
265 float out, in = src[i];
266
267 out = in * s->in_gain;
268
269 for (n = 0; n < s->num_chorus; n++) {
270 out += chorusbuf[MOD(s->max_samples + s->counter[c] -
271 s->lookup_table[n][phase[n]],
272 s->max_samples)] * s->decays[n];
273 phase[n] = MOD(phase[n] + 1, s->length[n]);
274 }
275
276 out *= s->out_gain;
277
278 dst[i] = out;
279
280 chorusbuf[s->counter[c]] = in;
281 s->counter[c] = MOD(s->counter[c] + 1, s->max_samples);
282 }
283 }
284
285 s->next_pts = frame->pts + av_rescale_q(frame->nb_samples, (AVRational){1, inlink->sample_rate}, inlink->time_base);
286
287 if (frame != out_frame)
288 av_frame_free(&frame);
289
290 return ff_filter_frame(ctx->outputs[0], out_frame);
291}
292
293static int request_frame(AVFilterLink *outlink)
294{
295 AVFilterContext *ctx = outlink->src;
296 ChorusContext *s = ctx->priv;
297 int ret;
298
299 ret = ff_request_frame(ctx->inputs[0]);
300
301 if (ret == AVERROR_EOF && !ctx->is_disabled && s->fade_out) {
302 int nb_samples = FFMIN(s->fade_out, 2048);
303 AVFrame *frame;
304
305 frame = ff_get_audio_buffer(outlink, nb_samples);
306 if (!frame)
307 return AVERROR(ENOMEM);
308 s->fade_out -= nb_samples;
309
310 av_samples_set_silence(frame->extended_data, 0,
311 frame->nb_samples,
312 outlink->channels,
313 frame->format);
314
315 frame->pts = s->next_pts;
316 if (s->next_pts != AV_NOPTS_VALUE)
317 s->next_pts += av_rescale_q(nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base);
318
319 ret = filter_frame(ctx->inputs[0], frame);
320 }
321
322 return ret;
323}
324
325static av_cold void uninit(AVFilterContext *ctx)
326{
327 ChorusContext *s = ctx->priv;
328 int n;
329
330 av_freep(&s->delays);
331 av_freep(&s->decays);
332 av_freep(&s->speeds);
333 av_freep(&s->depths);
334
335 if (s->chorusbuf)
336 av_freep(&s->chorusbuf[0]);
337 av_freep(&s->chorusbuf);
338
339 if (s->phase)
340 for (n = 0; n < s->channels; n++)
341 av_freep(&s->phase[n]);
342 av_freep(&s->phase);
343
344 av_freep(&s->counter);
345 av_freep(&s->length);
346
347 if (s->lookup_table)
348 for (n = 0; n < s->num_chorus; n++)
349 av_freep(&s->lookup_table[n]);
350 av_freep(&s->lookup_table);
351}
352
353static const AVFilterPad chorus_inputs[] = {
354 {
355 .name = "default",
356 .type = AVMEDIA_TYPE_AUDIO,
357 .filter_frame = filter_frame,
358 },
359 { NULL }
360};
361
362static const AVFilterPad chorus_outputs[] = {
363 {
364 .name = "default",
365 .type = AVMEDIA_TYPE_AUDIO,
366 .request_frame = request_frame,
367 .config_props = config_output,
368 },
369 { NULL }
370};
371
372AVFilter ff_af_chorus = {
373 .name = "chorus",
374 .description = NULL_IF_CONFIG_SMALL("Add a chorus effect to the audio."),
375 .query_formats = query_formats,
376 .priv_size = sizeof(ChorusContext),
377 .priv_class = &chorus_class,
378 .init = init,
379 .uninit = uninit,
380 .inputs = chorus_inputs,
381 .outputs = chorus_outputs,
382};
383
generated by cgit at 2024-05-17 06:33:29 (GMT)