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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 #include "generate_wave_table.h"
32 
33 typedef struct AudioPhaserContext {
34  const AVClass *class;
35  double in_gain, out_gain;
36  double delay;
37  double decay;
38  double speed;
39 
40  enum WaveType type;
41 
43  double *delay_buffer;
44 
47 
49 
51  uint8_t * const *src, uint8_t **dst,
52  int nb_samples, int channels);
54 
55 #define OFFSET(x) offsetof(AudioPhaserContext, x)
56 #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
57 
58 static const AVOption aphaser_options[] = {
59  { "in_gain", "set input gain", OFFSET(in_gain), AV_OPT_TYPE_DOUBLE, {.dbl=.4}, 0, 1, FLAGS },
60  { "out_gain", "set output gain", OFFSET(out_gain), AV_OPT_TYPE_DOUBLE, {.dbl=.74}, 0, 1e9, FLAGS },
61  { "delay", "set delay in milliseconds", OFFSET(delay), AV_OPT_TYPE_DOUBLE, {.dbl=3.}, 0, 5, FLAGS },
62  { "decay", "set decay", OFFSET(decay), AV_OPT_TYPE_DOUBLE, {.dbl=.4}, 0, .99, FLAGS },
63  { "speed", "set modulation speed", OFFSET(speed), AV_OPT_TYPE_DOUBLE, {.dbl=.5}, .1, 2, FLAGS },
64  { "type", "set modulation type", OFFSET(type), AV_OPT_TYPE_INT, {.i64=WAVE_TRI}, 0, WAVE_NB-1, FLAGS, "type" },
65  { "triangular", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_TRI}, 0, 0, FLAGS, "type" },
66  { "t", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_TRI}, 0, 0, FLAGS, "type" },
67  { "sinusoidal", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_SIN}, 0, 0, FLAGS, "type" },
68  { "s", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_SIN}, 0, 0, FLAGS, "type" },
69  { NULL }
70 };
71 
72 AVFILTER_DEFINE_CLASS(aphaser);
73 
74 static av_cold int init(AVFilterContext *ctx)
75 {
76  AudioPhaserContext *p = ctx->priv;
77 
78  if (p->in_gain > (1 - p->decay * p->decay))
79  av_log(ctx, AV_LOG_WARNING, "in_gain may cause clipping\n");
80  if (p->in_gain / (1 - p->decay) > 1 / p->out_gain)
81  av_log(ctx, AV_LOG_WARNING, "out_gain may cause clipping\n");
82 
83  return 0;
84 }
85 
87 {
90  static const enum AVSampleFormat sample_fmts[] = {
96  };
97 
98  layouts = ff_all_channel_layouts();
99  if (!layouts)
100  return AVERROR(ENOMEM);
101  ff_set_common_channel_layouts(ctx, layouts);
102 
103  formats = ff_make_format_list(sample_fmts);
104  if (!formats)
105  return AVERROR(ENOMEM);
106  ff_set_common_formats(ctx, formats);
107 
108  formats = ff_all_samplerates();
109  if (!formats)
110  return AVERROR(ENOMEM);
111  ff_set_common_samplerates(ctx, formats);
112 
113  return 0;
114 }
115 
116 #define MOD(a, b) (((a) >= (b)) ? (a) - (b) : (a))
117 
118 #define PHASER_PLANAR(name, type) \
119 static void phaser_## name ##p(AudioPhaserContext *p, \
120  uint8_t * const *src, uint8_t **dst, \
121  int nb_samples, int channels) \
122 { \
123  int i, c, delay_pos, modulation_pos; \
124  \
125  av_assert0(channels > 0); \
126  for (c = 0; c < channels; c++) { \
127  type *s = (type *)src[c]; \
128  type *d = (type *)dst[c]; \
129  double *buffer = p->delay_buffer + \
130  c * p->delay_buffer_length; \
131  \
132  delay_pos = p->delay_pos; \
133  modulation_pos = p->modulation_pos; \
134  \
135  for (i = 0; i < nb_samples; i++, s++, d++) { \
136  double v = *s * p->in_gain + buffer[ \
137  MOD(delay_pos + p->modulation_buffer[ \
138  modulation_pos], \
139  p->delay_buffer_length)] * p->decay; \
140  \
141  modulation_pos = MOD(modulation_pos + 1, \
142  p->modulation_buffer_length); \
143  delay_pos = MOD(delay_pos + 1, p->delay_buffer_length); \
144  buffer[delay_pos] = v; \
145  \
146  *d = v * p->out_gain; \
147  } \
148  } \
149  \
150  p->delay_pos = delay_pos; \
151  p->modulation_pos = modulation_pos; \
152 }
153 
154 #define PHASER(name, type) \
155 static void phaser_## name (AudioPhaserContext *p, \
156  uint8_t * const *src, uint8_t **dst, \
157  int nb_samples, int channels) \
158 { \
159  int i, c, delay_pos, modulation_pos; \
160  type *s = (type *)src[0]; \
161  type *d = (type *)dst[0]; \
162  double *buffer = p->delay_buffer; \
163  \
164  delay_pos = p->delay_pos; \
165  modulation_pos = p->modulation_pos; \
166  \
167  for (i = 0; i < nb_samples; i++) { \
168  int pos = MOD(delay_pos + p->modulation_buffer[modulation_pos], \
169  p->delay_buffer_length) * channels; \
170  int npos; \
171  \
172  delay_pos = MOD(delay_pos + 1, p->delay_buffer_length); \
173  npos = delay_pos * channels; \
174  for (c = 0; c < channels; c++, s++, d++) { \
175  double v = *s * p->in_gain + buffer[pos + c] * p->decay; \
176  \
177  buffer[npos + c] = v; \
178  \
179  *d = v * p->out_gain; \
180  } \
181  \
182  modulation_pos = MOD(modulation_pos + 1, \
183  p->modulation_buffer_length); \
184  } \
185  \
186  p->delay_pos = delay_pos; \
187  p->modulation_pos = modulation_pos; \
188 }
189 
190 PHASER_PLANAR(dbl, double)
191 PHASER_PLANAR(flt, float)
192 PHASER_PLANAR(s16, int16_t)
194 
195 PHASER(dbl, double)
196 PHASER(flt, float)
197 PHASER(s16, int16_t)
198 PHASER(s32, int32_t)
199 
200 static int config_output(AVFilterLink *outlink)
201 {
202  AudioPhaserContext *p = outlink->src->priv;
203  AVFilterLink *inlink = outlink->src->inputs[0];
204 
205  p->delay_buffer_length = p->delay * 0.001 * inlink->sample_rate + 0.5;
206  p->delay_buffer = av_calloc(p->delay_buffer_length, sizeof(*p->delay_buffer) * inlink->channels);
207  p->modulation_buffer_length = inlink->sample_rate / p->speed + 0.5;
209 
210  if (!p->modulation_buffer || !p->delay_buffer)
211  return AVERROR(ENOMEM);
212 
215  1., p->delay_buffer_length, M_PI / 2.0);
216 
217  p->delay_pos = p->modulation_pos = 0;
218 
219  switch (inlink->format) {
220  case AV_SAMPLE_FMT_DBL: p->phaser = phaser_dbl; break;
221  case AV_SAMPLE_FMT_DBLP: p->phaser = phaser_dblp; break;
222  case AV_SAMPLE_FMT_FLT: p->phaser = phaser_flt; break;
223  case AV_SAMPLE_FMT_FLTP: p->phaser = phaser_fltp; break;
224  case AV_SAMPLE_FMT_S16: p->phaser = phaser_s16; break;
225  case AV_SAMPLE_FMT_S16P: p->phaser = phaser_s16p; break;
226  case AV_SAMPLE_FMT_S32: p->phaser = phaser_s32; break;
227  case AV_SAMPLE_FMT_S32P: p->phaser = phaser_s32p; break;
228  default: av_assert0(0);
229  }
230 
231  return 0;
232 }
233 
234 static int filter_frame(AVFilterLink *inlink, AVFrame *inbuf)
235 {
236  AudioPhaserContext *p = inlink->dst->priv;
237  AVFilterLink *outlink = inlink->dst->outputs[0];
238  AVFrame *outbuf;
239 
240  if (av_frame_is_writable(inbuf)) {
241  outbuf = inbuf;
242  } else {
243  outbuf = ff_get_audio_buffer(inlink, inbuf->nb_samples);
244  if (!outbuf)
245  return AVERROR(ENOMEM);
246  av_frame_copy_props(outbuf, inbuf);
247  }
248 
249  p->phaser(p, inbuf->extended_data, outbuf->extended_data,
250  outbuf->nb_samples, av_frame_get_channels(outbuf));
251 
252  if (inbuf != outbuf)
253  av_frame_free(&inbuf);
254 
255  return ff_filter_frame(outlink, outbuf);
256 }
257 
258 static av_cold void uninit(AVFilterContext *ctx)
259 {
260  AudioPhaserContext *p = ctx->priv;
261 
262  av_freep(&p->delay_buffer);
264 }
265 
266 static const AVFilterPad aphaser_inputs[] = {
267  {
268  .name = "default",
269  .type = AVMEDIA_TYPE_AUDIO,
270  .filter_frame = filter_frame,
271  },
272  { NULL }
273 };
274 
275 static const AVFilterPad aphaser_outputs[] = {
276  {
277  .name = "default",
278  .type = AVMEDIA_TYPE_AUDIO,
279  .config_props = config_output,
280  },
281  { NULL }
282 };
283 
285  .name = "aphaser",
286  .description = NULL_IF_CONFIG_SMALL("Add a phasing effect to the audio."),
287  .query_formats = query_formats,
288  .priv_size = sizeof(AudioPhaserContext),
289  .init = init,
290  .uninit = uninit,
291  .inputs = aphaser_inputs,
292  .outputs = aphaser_outputs,
293  .priv_class = &aphaser_class,
294 };