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af_aphaser.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 /**
22  * @file
23  * phaser audio filter
24  */
25 
26 #include "libavutil/avassert.h"
27 #include "libavutil/opt.h"
28 #include "audio.h"
29 #include "avfilter.h"
30 #include "internal.h"
31 
32 enum WaveType {
36 };
37 
38 typedef struct AudioPhaserContext {
39  const AVClass *class;
40  double in_gain, out_gain;
41  double delay;
42  double decay;
43  double speed;
44 
45  enum WaveType type;
46 
48  double *delay_buffer;
49 
52 
54 
56  uint8_t * const *src, uint8_t **dst,
57  int nb_samples, int channels);
59 
60 #define OFFSET(x) offsetof(AudioPhaserContext, x)
61 #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
62 
63 static const AVOption aphaser_options[] = {
64  { "in_gain", "set input gain", OFFSET(in_gain), AV_OPT_TYPE_DOUBLE, {.dbl=.4}, 0, 1, FLAGS },
65  { "out_gain", "set output gain", OFFSET(out_gain), AV_OPT_TYPE_DOUBLE, {.dbl=.74}, 0, 1e9, FLAGS },
66  { "delay", "set delay in milliseconds", OFFSET(delay), AV_OPT_TYPE_DOUBLE, {.dbl=3.}, 0, 5, FLAGS },
67  { "decay", "set decay", OFFSET(decay), AV_OPT_TYPE_DOUBLE, {.dbl=.4}, 0, .99, FLAGS },
68  { "speed", "set modulation speed", OFFSET(speed), AV_OPT_TYPE_DOUBLE, {.dbl=.5}, .1, 2, FLAGS },
69  { "type", "set modulation type", OFFSET(type), AV_OPT_TYPE_INT, {.i64=WAVE_TRI}, 0, WAVE_NB-1, FLAGS, "type" },
70  { "triangular", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_TRI}, 0, 0, FLAGS, "type" },
71  { "t", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_TRI}, 0, 0, FLAGS, "type" },
72  { "sinusoidal", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_SIN}, 0, 0, FLAGS, "type" },
73  { "s", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_SIN}, 0, 0, FLAGS, "type" },
74  { NULL }
75 };
76 
77 AVFILTER_DEFINE_CLASS(aphaser);
78 
79 static av_cold int init(AVFilterContext *ctx)
80 {
81  AudioPhaserContext *p = ctx->priv;
82 
83  if (p->in_gain > (1 - p->decay * p->decay))
84  av_log(ctx, AV_LOG_WARNING, "in_gain may cause clipping\n");
85  if (p->in_gain / (1 - p->decay) > 1 / p->out_gain)
86  av_log(ctx, AV_LOG_WARNING, "out_gain may cause clipping\n");
87 
88  return 0;
89 }
90 
92 {
95  static const enum AVSampleFormat sample_fmts[] = {
101  };
102 
103  layouts = ff_all_channel_layouts();
104  if (!layouts)
105  return AVERROR(ENOMEM);
106  ff_set_common_channel_layouts(ctx, layouts);
107 
108  formats = ff_make_format_list(sample_fmts);
109  if (!formats)
110  return AVERROR(ENOMEM);
111  ff_set_common_formats(ctx, formats);
112 
113  formats = ff_all_samplerates();
114  if (!formats)
115  return AVERROR(ENOMEM);
116  ff_set_common_samplerates(ctx, formats);
117 
118  return 0;
119 }
120 
121 static void generate_wave_table(enum WaveType wave_type, enum AVSampleFormat sample_fmt,
122  void *table, int table_size,
123  double min, double max, double phase)
124 {
125  uint32_t i, phase_offset = phase / M_PI / 2 * table_size + 0.5;
126 
127  for (i = 0; i < table_size; i++) {
128  uint32_t point = (i + phase_offset) % table_size;
129  double d;
130 
131  switch (wave_type) {
132  case WAVE_SIN:
133  d = (sin((double)point / table_size * 2 * M_PI) + 1) / 2;
134  break;
135  case WAVE_TRI:
136  d = (double)point * 2 / table_size;
137  switch (4 * point / table_size) {
138  case 0: d = d + 0.5; break;
139  case 1:
140  case 2: d = 1.5 - d; break;
141  case 3: d = d - 1.5; break;
142  }
143  break;
144  default:
145  av_assert0(0);
146  }
147 
148  d = d * (max - min) + min;
149  switch (sample_fmt) {
150  case AV_SAMPLE_FMT_FLT: {
151  float *fp = (float *)table;
152  *fp++ = (float)d;
153  table = fp;
154  continue; }
155  case AV_SAMPLE_FMT_DBL: {
156  double *dp = (double *)table;
157  *dp++ = d;
158  table = dp;
159  continue; }
160  }
161 
162  d += d < 0 ? -0.5 : 0.5;
163  switch (sample_fmt) {
164  case AV_SAMPLE_FMT_S16: {
165  int16_t *sp = table;
166  *sp++ = (int16_t)d;
167  table = sp;
168  continue; }
169  case AV_SAMPLE_FMT_S32: {
170  int32_t *ip = table;
171  *ip++ = (int32_t)d;
172  table = ip;
173  continue; }
174  default:
175  av_assert0(0);
176  }
177  }
178 }
179 
180 #define MOD(a, b) (((a) >= (b)) ? (a) - (b) : (a))
181 
182 #define PHASER_PLANAR(name, type) \
183 static void phaser_## name ##p(AudioPhaserContext *p, \
184  uint8_t * const *src, uint8_t **dst, \
185  int nb_samples, int channels) \
186 { \
187  int i, c, delay_pos, modulation_pos; \
188  \
189  av_assert0(channels > 0); \
190  for (c = 0; c < channels; c++) { \
191  type *s = (type *)src[c]; \
192  type *d = (type *)dst[c]; \
193  double *buffer = p->delay_buffer + \
194  c * p->delay_buffer_length; \
195  \
196  delay_pos = p->delay_pos; \
197  modulation_pos = p->modulation_pos; \
198  \
199  for (i = 0; i < nb_samples; i++, s++, d++) { \
200  double v = *s * p->in_gain + buffer[ \
201  MOD(delay_pos + p->modulation_buffer[ \
202  modulation_pos], \
203  p->delay_buffer_length)] * p->decay; \
204  \
205  modulation_pos = MOD(modulation_pos + 1, \
206  p->modulation_buffer_length); \
207  delay_pos = MOD(delay_pos + 1, p->delay_buffer_length); \
208  buffer[delay_pos] = v; \
209  \
210  *d = v * p->out_gain; \
211  } \
212  } \
213  \
214  p->delay_pos = delay_pos; \
215  p->modulation_pos = modulation_pos; \
216 }
217 
218 #define PHASER(name, type) \
219 static void phaser_## name (AudioPhaserContext *p, \
220  uint8_t * const *src, uint8_t **dst, \
221  int nb_samples, int channels) \
222 { \
223  int i, c, delay_pos, modulation_pos; \
224  type *s = (type *)src[0]; \
225  type *d = (type *)dst[0]; \
226  double *buffer = p->delay_buffer; \
227  \
228  delay_pos = p->delay_pos; \
229  modulation_pos = p->modulation_pos; \
230  \
231  for (i = 0; i < nb_samples; i++) { \
232  int pos = MOD(delay_pos + p->modulation_buffer[modulation_pos], \
233  p->delay_buffer_length) * channels; \
234  int npos; \
235  \
236  delay_pos = MOD(delay_pos + 1, p->delay_buffer_length); \
237  npos = delay_pos * channels; \
238  for (c = 0; c < channels; c++, s++, d++) { \
239  double v = *s * p->in_gain + buffer[pos + c] * p->decay; \
240  \
241  buffer[npos + c] = v; \
242  \
243  *d = v * p->out_gain; \
244  } \
245  \
246  modulation_pos = MOD(modulation_pos + 1, \
247  p->modulation_buffer_length); \
248  } \
249  \
250  p->delay_pos = delay_pos; \
251  p->modulation_pos = modulation_pos; \
252 }
253 
254 PHASER_PLANAR(dbl, double)
255 PHASER_PLANAR(flt, float)
256 PHASER_PLANAR(s16, int16_t)
258 
259 PHASER(dbl, double)
260 PHASER(flt, float)
261 PHASER(s16, int16_t)
262 PHASER(s32, int32_t)
263 
264 static int config_output(AVFilterLink *outlink)
265 {
266  AudioPhaserContext *p = outlink->src->priv;
267  AVFilterLink *inlink = outlink->src->inputs[0];
268 
269  p->delay_buffer_length = p->delay * 0.001 * inlink->sample_rate + 0.5;
270  p->delay_buffer = av_calloc(p->delay_buffer_length, sizeof(*p->delay_buffer) * inlink->channels);
271  p->modulation_buffer_length = inlink->sample_rate / p->speed + 0.5;
273 
274  if (!p->modulation_buffer || !p->delay_buffer)
275  return AVERROR(ENOMEM);
276 
279  1., p->delay_buffer_length, M_PI / 2.0);
280 
281  p->delay_pos = p->modulation_pos = 0;
282 
283  switch (inlink->format) {
284  case AV_SAMPLE_FMT_DBL: p->phaser = phaser_dbl; break;
285  case AV_SAMPLE_FMT_DBLP: p->phaser = phaser_dblp; break;
286  case AV_SAMPLE_FMT_FLT: p->phaser = phaser_flt; break;
287  case AV_SAMPLE_FMT_FLTP: p->phaser = phaser_fltp; break;
288  case AV_SAMPLE_FMT_S16: p->phaser = phaser_s16; break;
289  case AV_SAMPLE_FMT_S16P: p->phaser = phaser_s16p; break;
290  case AV_SAMPLE_FMT_S32: p->phaser = phaser_s32; break;
291  case AV_SAMPLE_FMT_S32P: p->phaser = phaser_s32p; break;
292  default: av_assert0(0);
293  }
294 
295  return 0;
296 }
297 
298 static int filter_frame(AVFilterLink *inlink, AVFrame *inbuf)
299 {
300  AudioPhaserContext *p = inlink->dst->priv;
301  AVFilterLink *outlink = inlink->dst->outputs[0];
302  AVFrame *outbuf;
303 
304  if (av_frame_is_writable(inbuf)) {
305  outbuf = inbuf;
306  } else {
307  outbuf = ff_get_audio_buffer(inlink, inbuf->nb_samples);
308  if (!outbuf)
309  return AVERROR(ENOMEM);
310  av_frame_copy_props(outbuf, inbuf);
311  }
312 
313  p->phaser(p, inbuf->extended_data, outbuf->extended_data,
314  outbuf->nb_samples, av_frame_get_channels(outbuf));
315 
316  if (inbuf != outbuf)
317  av_frame_free(&inbuf);
318 
319  return ff_filter_frame(outlink, outbuf);
320 }
321 
322 static av_cold void uninit(AVFilterContext *ctx)
323 {
324  AudioPhaserContext *p = ctx->priv;
325 
326  av_freep(&p->delay_buffer);
328 }
329 
330 static const AVFilterPad aphaser_inputs[] = {
331  {
332  .name = "default",
333  .type = AVMEDIA_TYPE_AUDIO,
334  .filter_frame = filter_frame,
335  },
336  { NULL }
337 };
338 
339 static const AVFilterPad aphaser_outputs[] = {
340  {
341  .name = "default",
342  .type = AVMEDIA_TYPE_AUDIO,
343  .config_props = config_output,
344  },
345  { NULL }
346 };
347 
349  .name = "aphaser",
350  .description = NULL_IF_CONFIG_SMALL("Add a phasing effect to the audio."),
351  .query_formats = query_formats,
352  .priv_size = sizeof(AudioPhaserContext),
353  .init = init,
354  .uninit = uninit,
355  .inputs = aphaser_inputs,
356  .outputs = aphaser_outputs,
357  .priv_class = &aphaser_class,
358 };