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