FFmpeg coverage

Directory: ../../../ffmpeg/
File: src/libavfilter/af_aecho.c
Date: 2026-04-21 18:58:04
Exec Total Coverage
Lines: 106 138 76.8%
Functions: 9 12 75.0%
Branches: 56 117 47.9%
Line Branch Exec Source
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 #include "libavutil/avassert.h"
22 #include "libavutil/avstring.h"
23 #include "libavutil/mem.h"
24 #include "libavutil/opt.h"
25 #include "libavutil/samplefmt.h"
26 #include "avfilter.h"
27 #include "audio.h"
28 #include "filters.h"
29
30 typedef struct AudioEchoContext {
31 const AVClass *class;
32 float in_gain, out_gain;
33 char *delays, *decays;
34 float *delay, *decay;
35 int nb_echoes;
36 int delay_index;
37 uint8_t **delayptrs;
38 int max_samples, fade_out;
39 int *samples;
40 int eof;
41 int64_t next_pts;
42
43 void (*echo_samples)(struct AudioEchoContext *ctx, uint8_t **delayptrs,
44 uint8_t * const *src, uint8_t **dst,
45 int nb_samples, int channels);
46 } AudioEchoContext;
47
48 #define OFFSET(x) offsetof(AudioEchoContext, x)
49 #define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
50
51 static const AVOption aecho_options[] = {
52 { "in_gain", "set signal input gain", OFFSET(in_gain), AV_OPT_TYPE_FLOAT, {.dbl=0.6}, 0, 1, A },
53 { "out_gain", "set signal output gain", OFFSET(out_gain), AV_OPT_TYPE_FLOAT, {.dbl=0.3}, 0, 1, A },
54 { "delays", "set list of signal delays", OFFSET(delays), AV_OPT_TYPE_STRING, {.str="1000"}, 0, 0, A },
55 { "decays", "set list of signal decays", OFFSET(decays), AV_OPT_TYPE_STRING, {.str="0.5"}, 0, 0, A },
56 { NULL }
57 };
58
59 AVFILTER_DEFINE_CLASS(aecho);
60
61 4 static void count_items(char *item_str, int *nb_items)
62 {
63 char *p;
64
65 4 *nb_items = 1;
66
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 14 for (p = item_str; *p; p++) {
67
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 10 if (*p == '|')
68 ✗ (*nb_items)++;
69 }
70
71 4 }
72
73 4 static void fill_items(char *item_str, int *nb_items, float *items)
74 {
75 4 char *p, *saveptr = NULL;
76 4 int i, new_nb_items = 0;
77
78 4 p = item_str;
79
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 8 for (i = 0; i < *nb_items; i++) {
80 4 char *tstr = av_strtok(p, "|", &saveptr);
81 4 p = NULL;
82
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 4 if (tstr)
83 4 new_nb_items += av_sscanf(tstr, "%f", &items[new_nb_items]) == 1;
84 }
85
86 4 *nb_items = new_nb_items;
87 4 }
88
89 2 static av_cold void uninit(AVFilterContext *ctx)
90 {
91 2 AudioEchoContext *s = ctx->priv;
92
93 2 av_freep(&s->delay);
94 2 av_freep(&s->decay);
95 2 av_freep(&s->samples);
96
97
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 2 if (s->delayptrs)
98 1 av_freep(&s->delayptrs[0]);
99 2 av_freep(&s->delayptrs);
100 2 }
101
102 2 static av_cold int init(AVFilterContext *ctx)
103 {
104 2 AudioEchoContext *s = ctx->priv;
105 int nb_delays, nb_decays, i;
106
107
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 2 if (!s->delays || !s->decays) {
108 ✗ av_log(ctx, AV_LOG_ERROR, "Missing delays and/or decays.\n");
109 ✗ return AVERROR(EINVAL);
110 }
111
112 2 count_items(s->delays, &nb_delays);
113 2 count_items(s->decays, &nb_decays);
114
115 2 s->delay = av_realloc_f(s->delay, nb_delays, sizeof(*s->delay));
116 2 s->decay = av_realloc_f(s->decay, nb_decays, sizeof(*s->decay));
117
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 2 if (!s->delay || !s->decay)
118 ✗ return AVERROR(ENOMEM);
119
120 2 fill_items(s->delays, &nb_delays, s->delay);
121 2 fill_items(s->decays, &nb_decays, s->decay);
122
123
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 2 if (nb_delays != nb_decays) {
124 ✗ av_log(ctx, AV_LOG_ERROR, "Number of delays %d differs from number of decays %d.\n", nb_delays, nb_decays);
125 ✗ return AVERROR(EINVAL);
126 }
127
128 2 s->nb_echoes = nb_delays;
129
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 2 if (!s->nb_echoes) {
130 ✗ av_log(ctx, AV_LOG_ERROR, "At least one decay & delay must be set.\n");
131 ✗ return AVERROR(EINVAL);
132 }
133
134 2 s->samples = av_realloc_f(s->samples, nb_delays, sizeof(*s->samples));
135
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 2 if (!s->samples)
136 ✗ return AVERROR(ENOMEM);
137
138
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 4 for (i = 0; i < nb_delays; i++) {
139
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 2 if (s->delay[i] <= 0 || s->delay[i] > 90000) {
140 ✗ av_log(ctx, AV_LOG_ERROR, "delay[%d]: %f is out of allowed range: (0, 90000]\n", i, s->delay[i]);
141 ✗ return AVERROR(EINVAL);
142 }
143
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 2 if (s->decay[i] <= 0 || s->decay[i] > 1) {
144 ✗ av_log(ctx, AV_LOG_ERROR, "decay[%d]: %f is out of allowed range: (0, 1]\n", i, s->decay[i]);
145 ✗ return AVERROR(EINVAL);
146 }
147 }
148
149 2 s->next_pts = AV_NOPTS_VALUE;
150
151 2 av_log(ctx, AV_LOG_DEBUG, "nb_echoes:%d\n", s->nb_echoes);
152 2 return 0;
153 }
154
155 #define MOD(a, b) (((a) >= (b)) ? (a) - (b) : (a))
156
157 #define ECHO(name, type, min, max) \
158 static void echo_samples_## name ##p(AudioEchoContext *ctx, \
159 uint8_t **delayptrs, \
160 uint8_t * const *src, uint8_t **dst, \
161 int nb_samples, int channels) \
162 { \
163 const double out_gain = ctx->out_gain; \
164 const double in_gain = ctx->in_gain; \
165 const int nb_echoes = ctx->nb_echoes; \
166 const int max_samples = ctx->max_samples; \
167 int i, j, chan, av_uninit(index); \
168 \
169 av_assert1(channels > 0); /* would corrupt delay_index */ \
170 \
171 for (chan = 0; chan < channels; chan++) { \
172 const type *s = (type *)src[chan]; \
173 type *d = (type *)dst[chan]; \
174 type *dbuf = (type *)delayptrs[chan]; \
175 \
176 index = ctx->delay_index; \
177 for (i = 0; i < nb_samples; i++, s++, d++) { \
178 double out, in; \
179 \
180 in = *s; \
181 out = in * in_gain; \
182 for (j = 0; j < nb_echoes; j++) { \
183 int ix = index + max_samples - ctx->samples[j]; \
184 ix = MOD(ix, max_samples); \
185 out += dbuf[ix] * ctx->decay[j]; \
186 } \
187 out *= out_gain; \
188 \
189 *d = av_clipd(out, min, max); \
190 dbuf[index] = in; \
191 \
192 index = MOD(index + 1, max_samples); \
193 } \
194 } \
195 ctx->delay_index = index; \
196 }
197
198 ✗ ECHO(dbl, double, -1.0, 1.0 )
199 ✗ ECHO(flt, float, -1.0, 1.0 )
200
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 1064242 ECHO(s16, int16_t, INT16_MIN, INT16_MAX)
201 ✗ ECHO(s32, int32_t, INT32_MIN, INT32_MAX)
202
203 1 static int config_output(AVFilterLink *outlink)
204 {
205 1 AVFilterContext *ctx = outlink->src;
206 1 AudioEchoContext *s = ctx->priv;
207 1 float volume = 1.0;
208 int i;
209
210
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 2 for (i = 0; i < s->nb_echoes; i++) {
211 1 s->samples[i] = s->delay[i] * outlink->sample_rate / 1000.0;
212 1 s->max_samples = FFMAX(s->max_samples, s->samples[i]);
213 1 volume += s->decay[i];
214 }
215
216
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 1 if (s->max_samples <= 0) {
217 ✗ av_log(ctx, AV_LOG_ERROR, "Nothing to echo - missing delay samples.\n");
218 ✗ return AVERROR(EINVAL);
219 }
220 1 s->fade_out = s->max_samples;
221
222
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 1 if (volume * s->in_gain * s->out_gain > 1.0)
223 ✗ av_log(ctx, AV_LOG_WARNING,
224 ✗ "out_gain %f can cause saturation of output\n", s->out_gain);
225
226
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 1 switch (outlink->format) {
227 ✗ case AV_SAMPLE_FMT_DBLP: s->echo_samples = echo_samples_dblp; break;
228 ✗ case AV_SAMPLE_FMT_FLTP: s->echo_samples = echo_samples_fltp; break;
229 1 case AV_SAMPLE_FMT_S16P: s->echo_samples = echo_samples_s16p; break;
230 ✗ case AV_SAMPLE_FMT_S32P: s->echo_samples = echo_samples_s32p; break;
231 }
232
233
234
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 1 if (s->delayptrs)
235 ✗ av_freep(&s->delayptrs[0]);
236 1 av_freep(&s->delayptrs);
237
238 1 return av_samples_alloc_array_and_samples(&s->delayptrs, NULL,
239 outlink->ch_layout.nb_channels,
240 s->max_samples,
241 1 outlink->format, 0);
242 }
243
244 65 static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
245 {
246 65 AVFilterContext *ctx = inlink->dst;
247 65 AudioEchoContext *s = ctx->priv;
248 AVFrame *out_frame;
249
250
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 65 if (av_frame_is_writable(frame)) {
251 65 out_frame = frame;
252 } else {
253 ✗ out_frame = ff_get_audio_buffer(ctx->outputs[0], frame->nb_samples);
254 ✗ if (!out_frame) {
255 ✗ av_frame_free(&frame);
256 ✗ return AVERROR(ENOMEM);
257 }
258 ✗ av_frame_copy_props(out_frame, frame);
259 }
260
261 65 s->echo_samples(s, s->delayptrs, frame->extended_data, out_frame->extended_data,
262 65 frame->nb_samples, inlink->ch_layout.nb_channels);
263
264 65 s->next_pts = frame->pts + av_rescale_q(frame->nb_samples, (AVRational){1, inlink->sample_rate}, inlink->time_base);
265
266
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 65 if (frame != out_frame)
267 ✗ av_frame_free(&frame);
268
269 65 return ff_filter_frame(ctx->outputs[0], out_frame);
270 }
271
272 1 static int request_frame(AVFilterLink *outlink)
273 {
274 1 AVFilterContext *ctx = outlink->src;
275 1 AudioEchoContext *s = ctx->priv;
276 1 int nb_samples = FFMIN(s->fade_out, 2048);
277 1 AVFrame *frame = ff_get_audio_buffer(outlink, nb_samples);
278
279
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 1 if (!frame)
280 ✗ return AVERROR(ENOMEM);
281 1 s->fade_out -= nb_samples;
282
283 1 av_samples_set_silence(frame->extended_data, 0,
284 frame->nb_samples,
285 outlink->ch_layout.nb_channels,
286 1 frame->format);
287
288 1 s->echo_samples(s, s->delayptrs, frame->extended_data, frame->extended_data,
289 frame->nb_samples, outlink->ch_layout.nb_channels);
290
291 1 frame->pts = s->next_pts;
292
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 1 if (s->next_pts != AV_NOPTS_VALUE)
293 1 s->next_pts += av_rescale_q(nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base);
294
295 1 return ff_filter_frame(outlink, frame);
296 }
297
298 132 static int activate(AVFilterContext *ctx)
299 {
300 132 AVFilterLink *inlink = ctx->inputs[0];
301 132 AVFilterLink *outlink = ctx->outputs[0];
302 132 AudioEchoContext *s = ctx->priv;
303 AVFrame *in;
304 int ret, status;
305 int64_t pts;
306
307
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 132 FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink);
308
309 132 ret = ff_inlink_consume_frame(inlink, &in);
310
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 132 if (ret < 0)
311 ✗ return ret;
312
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 132 if (ret > 0)
313 65 return filter_frame(inlink, in);
314
315
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 67 if (!s->eof && ff_inlink_acknowledge_status(inlink, &status, &pts)) {
316
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 1 if (status == AVERROR_EOF)
317 1 s->eof = 1;
318 }
319
320
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 67 if (s->eof && s->fade_out <= 0) {
321 1 ff_outlink_set_status(outlink, AVERROR_EOF, s->next_pts);
322 1 return 0;
323 }
324
325
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 66 if (!s->eof)
326
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 65 FF_FILTER_FORWARD_WANTED(outlink, inlink);
327
328 1 return request_frame(outlink);
329 }
330
331 static const AVFilterPad aecho_outputs[] = {
332 {
333 .name = "default",
334 .config_props = config_output,
335 .type = AVMEDIA_TYPE_AUDIO,
336 },
337 };
338
339 const FFFilter ff_af_aecho = {
340 .p.name = "aecho",
341 .p.description = NULL_IF_CONFIG_SMALL("Add echoing to the audio."),
342 .p.priv_class = &aecho_class,
343 .priv_size = sizeof(AudioEchoContext),
344 .init = init,
345 .activate = activate,
346 .uninit = uninit,
347 FILTER_INPUTS(ff_audio_default_filterpad),
348 FILTER_OUTPUTS(aecho_outputs),
349 FILTER_SAMPLEFMTS(AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_S32P,
350 AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_DBLP),
351 };
352