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
39 typedef 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
66 static 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
76 AVFILTER_DEFINE_CLASS(chorus);
77
count_items(char * item_str,int * nb_items)78 static 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
fill_items(char * item_str,int * nb_items,float * items)90 static 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
init(AVFilterContext * ctx)106 static 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
config_output(AVFilterLink * outlink)157 static int config_output(AVFilterLink *outlink)
158 {
159 AVFilterContext *ctx = outlink->src;
160 ChorusContext *s = ctx->priv;
161 float sum_in_volume = 1.0;
162 int n;
163
164 s->channels = outlink->ch_layout.nb_channels;
165
166 for (n = 0; n < s->num_chorus; n++) {
167 int samples = (int) ((s->delays[n] + s->depths[n]) * outlink->sample_rate / 1000.0);
168 int depth_samples = (int) (s->depths[n] * outlink->sample_rate / 1000.0);
169
170 s->length[n] = outlink->sample_rate / s->speeds[n];
171
172 s->lookup_table[n] = av_malloc(sizeof(int32_t) * s->length[n]);
173 if (!s->lookup_table[n])
174 return AVERROR(ENOMEM);
175
176 ff_generate_wave_table(WAVE_SIN, AV_SAMPLE_FMT_S32, s->lookup_table[n],
177 s->length[n], 0., depth_samples, 0);
178 s->max_samples = FFMAX(s->max_samples, samples);
179 }
180
181 for (n = 0; n < s->num_chorus; n++)
182 sum_in_volume += s->decays[n];
183
184 if (s->in_gain * (sum_in_volume) > 1.0 / s->out_gain)
185 av_log(ctx, AV_LOG_WARNING, "output gain can cause saturation or clipping of output\n");
186
187 s->counter = av_calloc(outlink->ch_layout.nb_channels, sizeof(*s->counter));
188 if (!s->counter)
189 return AVERROR(ENOMEM);
190
191 s->phase = av_calloc(outlink->ch_layout.nb_channels, sizeof(*s->phase));
192 if (!s->phase)
193 return AVERROR(ENOMEM);
194
195 for (n = 0; n < outlink->ch_layout.nb_channels; n++) {
196 s->phase[n] = av_calloc(s->num_chorus, sizeof(int));
197 if (!s->phase[n])
198 return AVERROR(ENOMEM);
199 }
200
201 s->fade_out = s->max_samples;
202
203 return av_samples_alloc_array_and_samples(&s->chorusbuf, NULL,
204 outlink->ch_layout.nb_channels,
205 s->max_samples,
206 outlink->format, 0);
207 }
208
209 #define MOD(a, b) (((a) >= (b)) ? (a) - (b) : (a))
210
filter_frame(AVFilterLink * inlink,AVFrame * frame)211 static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
212 {
213 AVFilterContext *ctx = inlink->dst;
214 ChorusContext *s = ctx->priv;
215 AVFrame *out_frame;
216 int c, i, n;
217
218 if (av_frame_is_writable(frame)) {
219 out_frame = frame;
220 } else {
221 out_frame = ff_get_audio_buffer(ctx->outputs[0], frame->nb_samples);
222 if (!out_frame) {
223 av_frame_free(&frame);
224 return AVERROR(ENOMEM);
225 }
226 av_frame_copy_props(out_frame, frame);
227 }
228
229 for (c = 0; c < inlink->ch_layout.nb_channels; c++) {
230 const float *src = (const float *)frame->extended_data[c];
231 float *dst = (float *)out_frame->extended_data[c];
232 float *chorusbuf = (float *)s->chorusbuf[c];
233 int *phase = s->phase[c];
234
235 for (i = 0; i < frame->nb_samples; i++) {
236 float out, in = src[i];
237
238 out = in * s->in_gain;
239
240 for (n = 0; n < s->num_chorus; n++) {
241 out += chorusbuf[MOD(s->max_samples + s->counter[c] -
242 s->lookup_table[n][phase[n]],
243 s->max_samples)] * s->decays[n];
244 phase[n] = MOD(phase[n] + 1, s->length[n]);
245 }
246
247 out *= s->out_gain;
248
249 dst[i] = out;
250
251 chorusbuf[s->counter[c]] = in;
252 s->counter[c] = MOD(s->counter[c] + 1, s->max_samples);
253 }
254 }
255
256 s->next_pts = frame->pts + av_rescale_q(frame->nb_samples, (AVRational){1, inlink->sample_rate}, inlink->time_base);
257
258 if (frame != out_frame)
259 av_frame_free(&frame);
260
261 return ff_filter_frame(ctx->outputs[0], out_frame);
262 }
263
request_frame(AVFilterLink * outlink)264 static int request_frame(AVFilterLink *outlink)
265 {
266 AVFilterContext *ctx = outlink->src;
267 ChorusContext *s = ctx->priv;
268 int ret;
269
270 ret = ff_request_frame(ctx->inputs[0]);
271
272 if (ret == AVERROR_EOF && !ctx->is_disabled && s->fade_out) {
273 int nb_samples = FFMIN(s->fade_out, 2048);
274 AVFrame *frame;
275
276 frame = ff_get_audio_buffer(outlink, nb_samples);
277 if (!frame)
278 return AVERROR(ENOMEM);
279 s->fade_out -= nb_samples;
280
281 av_samples_set_silence(frame->extended_data, 0,
282 frame->nb_samples,
283 outlink->ch_layout.nb_channels,
284 frame->format);
285
286 frame->pts = s->next_pts;
287 if (s->next_pts != AV_NOPTS_VALUE)
288 s->next_pts += av_rescale_q(nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base);
289
290 ret = filter_frame(ctx->inputs[0], frame);
291 }
292
293 return ret;
294 }
295
uninit(AVFilterContext * ctx)296 static av_cold void uninit(AVFilterContext *ctx)
297 {
298 ChorusContext *s = ctx->priv;
299 int n;
300
301 av_freep(&s->delays);
302 av_freep(&s->decays);
303 av_freep(&s->speeds);
304 av_freep(&s->depths);
305
306 if (s->chorusbuf)
307 av_freep(&s->chorusbuf[0]);
308 av_freep(&s->chorusbuf);
309
310 if (s->phase)
311 for (n = 0; n < s->channels; n++)
312 av_freep(&s->phase[n]);
313 av_freep(&s->phase);
314
315 av_freep(&s->counter);
316 av_freep(&s->length);
317
318 if (s->lookup_table)
319 for (n = 0; n < s->num_chorus; n++)
320 av_freep(&s->lookup_table[n]);
321 av_freep(&s->lookup_table);
322 }
323
324 static const AVFilterPad chorus_inputs[] = {
325 {
326 .name = "default",
327 .type = AVMEDIA_TYPE_AUDIO,
328 .filter_frame = filter_frame,
329 },
330 };
331
332 static const AVFilterPad chorus_outputs[] = {
333 {
334 .name = "default",
335 .type = AVMEDIA_TYPE_AUDIO,
336 .request_frame = request_frame,
337 .config_props = config_output,
338 },
339 };
340
341 const AVFilter ff_af_chorus = {
342 .name = "chorus",
343 .description = NULL_IF_CONFIG_SMALL("Add a chorus effect to the audio."),
344 .priv_size = sizeof(ChorusContext),
345 .priv_class = &chorus_class,
346 .init = init,
347 .uninit = uninit,
348 FILTER_INPUTS(chorus_inputs),
349 FILTER_OUTPUTS(chorus_outputs),
350 FILTER_SINGLE_SAMPLEFMT(AV_SAMPLE_FMT_FLTP),
351 };
352