summaryrefslogtreecommitdiff
path: root/libavfilter/af_aecho.c (plain)
blob: cfaea3de43057fc2d37748f9e9e26813ce012de7
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/opt.h"
24#include "libavutil/samplefmt.h"
25#include "avfilter.h"
26#include "audio.h"
27#include "internal.h"
28
29typedef struct AudioEchoContext {
30 const AVClass *class;
31 float in_gain, out_gain;
32 char *delays, *decays;
33 float *delay, *decay;
34 int nb_echoes;
35 int delay_index;
36 uint8_t **delayptrs;
37 int max_samples, fade_out;
38 int *samples;
39 int64_t next_pts;
40
41 void (*echo_samples)(struct AudioEchoContext *ctx, uint8_t **delayptrs,
42 uint8_t * const *src, uint8_t **dst,
43 int nb_samples, int channels);
44} AudioEchoContext;
45
46#define OFFSET(x) offsetof(AudioEchoContext, x)
47#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
48
49static const AVOption aecho_options[] = {
50 { "in_gain", "set signal input gain", OFFSET(in_gain), AV_OPT_TYPE_FLOAT, {.dbl=0.6}, 0, 1, A },
51 { "out_gain", "set signal output gain", OFFSET(out_gain), AV_OPT_TYPE_FLOAT, {.dbl=0.3}, 0, 1, A },
52 { "delays", "set list of signal delays", OFFSET(delays), AV_OPT_TYPE_STRING, {.str="1000"}, 0, 0, A },
53 { "decays", "set list of signal decays", OFFSET(decays), AV_OPT_TYPE_STRING, {.str="0.5"}, 0, 0, A },
54 { NULL }
55};
56
57AVFILTER_DEFINE_CLASS(aecho);
58
59static void count_items(char *item_str, int *nb_items)
60{
61 char *p;
62
63 *nb_items = 1;
64 for (p = item_str; *p; p++) {
65 if (*p == '|')
66 (*nb_items)++;
67 }
68
69}
70
71static void fill_items(char *item_str, int *nb_items, float *items)
72{
73 char *p, *saveptr = NULL;
74 int i, new_nb_items = 0;
75
76 p = item_str;
77 for (i = 0; i < *nb_items; i++) {
78 char *tstr = av_strtok(p, "|", &saveptr);
79 p = NULL;
80 if (tstr)
81 new_nb_items += sscanf(tstr, "%f", &items[new_nb_items]) == 1;
82 }
83
84 *nb_items = new_nb_items;
85}
86
87static av_cold void uninit(AVFilterContext *ctx)
88{
89 AudioEchoContext *s = ctx->priv;
90
91 av_freep(&s->delay);
92 av_freep(&s->decay);
93 av_freep(&s->samples);
94
95 if (s->delayptrs)
96 av_freep(&s->delayptrs[0]);
97 av_freep(&s->delayptrs);
98}
99
100static av_cold int init(AVFilterContext *ctx)
101{
102 AudioEchoContext *s = ctx->priv;
103 int nb_delays, nb_decays, i;
104
105 if (!s->delays || !s->decays) {
106 av_log(ctx, AV_LOG_ERROR, "Missing delays and/or decays.\n");
107 return AVERROR(EINVAL);
108 }
109
110 count_items(s->delays, &nb_delays);
111 count_items(s->decays, &nb_decays);
112
113 s->delay = av_realloc_f(s->delay, nb_delays, sizeof(*s->delay));
114 s->decay = av_realloc_f(s->decay, nb_decays, sizeof(*s->decay));
115 if (!s->delay || !s->decay)
116 return AVERROR(ENOMEM);
117
118 fill_items(s->delays, &nb_delays, s->delay);
119 fill_items(s->decays, &nb_decays, s->decay);
120
121 if (nb_delays != nb_decays) {
122 av_log(ctx, AV_LOG_ERROR, "Number of delays %d differs from number of decays %d.\n", nb_delays, nb_decays);
123 return AVERROR(EINVAL);
124 }
125
126 s->nb_echoes = nb_delays;
127 if (!s->nb_echoes) {
128 av_log(ctx, AV_LOG_ERROR, "At least one decay & delay must be set.\n");
129 return AVERROR(EINVAL);
130 }
131
132 s->samples = av_realloc_f(s->samples, nb_delays, sizeof(*s->samples));
133 if (!s->samples)
134 return AVERROR(ENOMEM);
135
136 for (i = 0; i < nb_delays; i++) {
137 if (s->delay[i] <= 0 || s->delay[i] > 90000) {
138 av_log(ctx, AV_LOG_ERROR, "delay[%d]: %f is out of allowed range: (0, 90000]\n", i, s->delay[i]);
139 return AVERROR(EINVAL);
140 }
141 if (s->decay[i] <= 0 || s->decay[i] > 1) {
142 av_log(ctx, AV_LOG_ERROR, "decay[%d]: %f is out of allowed range: (0, 1]\n", i, s->decay[i]);
143 return AVERROR(EINVAL);
144 }
145 }
146
147 s->next_pts = AV_NOPTS_VALUE;
148
149 av_log(ctx, AV_LOG_DEBUG, "nb_echoes:%d\n", s->nb_echoes);
150 return 0;
151}
152
153static int query_formats(AVFilterContext *ctx)
154{
155 AVFilterChannelLayouts *layouts;
156 AVFilterFormats *formats;
157 static const enum AVSampleFormat sample_fmts[] = {
158 AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_S32P,
159 AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_DBLP,
160 AV_SAMPLE_FMT_NONE
161 };
162 int ret;
163
164 layouts = ff_all_channel_counts();
165 if (!layouts)
166 return AVERROR(ENOMEM);
167 ret = ff_set_common_channel_layouts(ctx, layouts);
168 if (ret < 0)
169 return ret;
170
171 formats = ff_make_format_list(sample_fmts);
172 if (!formats)
173 return AVERROR(ENOMEM);
174 ret = ff_set_common_formats(ctx, formats);
175 if (ret < 0)
176 return ret;
177
178 formats = ff_all_samplerates();
179 if (!formats)
180 return AVERROR(ENOMEM);
181 return ff_set_common_samplerates(ctx, formats);
182}
183
184#define MOD(a, b) (((a) >= (b)) ? (a) - (b) : (a))
185
186#define ECHO(name, type, min, max) \
187static void echo_samples_## name ##p(AudioEchoContext *ctx, \
188 uint8_t **delayptrs, \
189 uint8_t * const *src, uint8_t **dst, \
190 int nb_samples, int channels) \
191{ \
192 const double out_gain = ctx->out_gain; \
193 const double in_gain = ctx->in_gain; \
194 const int nb_echoes = ctx->nb_echoes; \
195 const int max_samples = ctx->max_samples; \
196 int i, j, chan, av_uninit(index); \
197 \
198 av_assert1(channels > 0); /* would corrupt delay_index */ \
199 \
200 for (chan = 0; chan < channels; chan++) { \
201 const type *s = (type *)src[chan]; \
202 type *d = (type *)dst[chan]; \
203 type *dbuf = (type *)delayptrs[chan]; \
204 \
205 index = ctx->delay_index; \
206 for (i = 0; i < nb_samples; i++, s++, d++) { \
207 double out, in; \
208 \
209 in = *s; \
210 out = in * in_gain; \
211 for (j = 0; j < nb_echoes; j++) { \
212 int ix = index + max_samples - ctx->samples[j]; \
213 ix = MOD(ix, max_samples); \
214 out += dbuf[ix] * ctx->decay[j]; \
215 } \
216 out *= out_gain; \
217 \
218 *d = av_clipd(out, min, max); \
219 dbuf[index] = in; \
220 \
221 index = MOD(index + 1, max_samples); \
222 } \
223 } \
224 ctx->delay_index = index; \
225}
226
227ECHO(dbl, double, -1.0, 1.0 )
228ECHO(flt, float, -1.0, 1.0 )
229ECHO(s16, int16_t, INT16_MIN, INT16_MAX)
230ECHO(s32, int32_t, INT32_MIN, INT32_MAX)
231
232static int config_output(AVFilterLink *outlink)
233{
234 AVFilterContext *ctx = outlink->src;
235 AudioEchoContext *s = ctx->priv;
236 float volume = 1.0;
237 int i;
238
239 for (i = 0; i < s->nb_echoes; i++) {
240 s->samples[i] = s->delay[i] * outlink->sample_rate / 1000.0;
241 s->max_samples = FFMAX(s->max_samples, s->samples[i]);
242 volume += s->decay[i];
243 }
244
245 if (s->max_samples <= 0) {
246 av_log(ctx, AV_LOG_ERROR, "Nothing to echo - missing delay samples.\n");
247 return AVERROR(EINVAL);
248 }
249 s->fade_out = s->max_samples;
250
251 if (volume * s->in_gain * s->out_gain > 1.0)
252 av_log(ctx, AV_LOG_WARNING,
253 "out_gain %f can cause saturation of output\n", s->out_gain);
254
255 switch (outlink->format) {
256 case AV_SAMPLE_FMT_DBLP: s->echo_samples = echo_samples_dblp; break;
257 case AV_SAMPLE_FMT_FLTP: s->echo_samples = echo_samples_fltp; break;
258 case AV_SAMPLE_FMT_S16P: s->echo_samples = echo_samples_s16p; break;
259 case AV_SAMPLE_FMT_S32P: s->echo_samples = echo_samples_s32p; break;
260 }
261
262
263 if (s->delayptrs)
264 av_freep(&s->delayptrs[0]);
265 av_freep(&s->delayptrs);
266
267 return av_samples_alloc_array_and_samples(&s->delayptrs, NULL,
268 outlink->channels,
269 s->max_samples,
270 outlink->format, 0);
271}
272
273static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
274{
275 AVFilterContext *ctx = inlink->dst;
276 AudioEchoContext *s = ctx->priv;
277 AVFrame *out_frame;
278
279 if (av_frame_is_writable(frame)) {
280 out_frame = frame;
281 } else {
282 out_frame = ff_get_audio_buffer(inlink, frame->nb_samples);
283 if (!out_frame) {
284 av_frame_free(&frame);
285 return AVERROR(ENOMEM);
286 }
287 av_frame_copy_props(out_frame, frame);
288 }
289
290 s->echo_samples(s, s->delayptrs, frame->extended_data, out_frame->extended_data,
291 frame->nb_samples, inlink->channels);
292
293 s->next_pts = frame->pts + av_rescale_q(frame->nb_samples, (AVRational){1, inlink->sample_rate}, inlink->time_base);
294
295 if (frame != out_frame)
296 av_frame_free(&frame);
297
298 return ff_filter_frame(ctx->outputs[0], out_frame);
299}
300
301static int request_frame(AVFilterLink *outlink)
302{
303 AVFilterContext *ctx = outlink->src;
304 AudioEchoContext *s = ctx->priv;
305 int ret;
306
307 ret = ff_request_frame(ctx->inputs[0]);
308
309 if (ret == AVERROR_EOF && !ctx->is_disabled && s->fade_out) {
310 int nb_samples = FFMIN(s->fade_out, 2048);
311 AVFrame *frame;
312
313 frame = ff_get_audio_buffer(outlink, nb_samples);
314 if (!frame)
315 return AVERROR(ENOMEM);
316 s->fade_out -= nb_samples;
317
318 av_samples_set_silence(frame->extended_data, 0,
319 frame->nb_samples,
320 outlink->channels,
321 frame->format);
322
323 s->echo_samples(s, s->delayptrs, frame->extended_data, frame->extended_data,
324 frame->nb_samples, outlink->channels);
325
326 frame->pts = s->next_pts;
327 if (s->next_pts != AV_NOPTS_VALUE)
328 s->next_pts += av_rescale_q(nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base);
329
330 return ff_filter_frame(outlink, frame);
331 }
332
333 return ret;
334}
335
336static const AVFilterPad aecho_inputs[] = {
337 {
338 .name = "default",
339 .type = AVMEDIA_TYPE_AUDIO,
340 .filter_frame = filter_frame,
341 },
342 { NULL }
343};
344
345static const AVFilterPad aecho_outputs[] = {
346 {
347 .name = "default",
348 .request_frame = request_frame,
349 .config_props = config_output,
350 .type = AVMEDIA_TYPE_AUDIO,
351 },
352 { NULL }
353};
354
355AVFilter ff_af_aecho = {
356 .name = "aecho",
357 .description = NULL_IF_CONFIG_SMALL("Add echoing to the audio."),
358 .query_formats = query_formats,
359 .priv_size = sizeof(AudioEchoContext),
360 .priv_class = &aecho_class,
361 .init = init,
362 .uninit = uninit,
363 .inputs = aecho_inputs,
364 .outputs = aecho_outputs,
365};
366