git.sesse.net Git - ffmpeg/blob - libavfilter/af_aphaser.c

   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 {
  33     WAVE_SIN,
  34     WAVE_TRI,
  35     WAVE_NB,
  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
  47     int delay_buffer_length;
  48     double *delay_buffer;
  49
  50     int modulation_buffer_length;
  51     int32_t *modulation_buffer;
  52
  53     int delay_pos, modulation_pos;
  54
  55     void (*phaser)(struct AudioPhaserContext *p,
  56                    uint8_t * const *src, uint8_t **dst,
  57                    int nb_samples, int channels);
  58 } AudioPhaserContext;
  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, const char *args)
  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
  91 static int query_formats(AVFilterContext *ctx)
  92 {
  93     AVFilterFormats *formats;
  94     AVFilterChannelLayouts *layouts;
  95     static const enum AVSampleFormat sample_fmts[] = {
  96         AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBLP,
  97         AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP,
  98         AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32P,
  99         AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16P,
 100         AV_SAMPLE_FMT_NONE
 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     for (c = 0; c < channels; c++) {                                   \
 190         type *s = (type *)src[c];                                      \
 191         type *d = (type *)dst[c];                                      \
 192         double *buffer = p->delay_buffer +                             \
 193                          c * p->delay_buffer_length;                   \
 194                                                                        \
 195         delay_pos      = p->delay_pos;                                 \
 196         modulation_pos = p->modulation_pos;                            \
 197                                                                        \
 198         for (i = 0; i < nb_samples; i++, s++, d++) {                   \
 199             double v = *s * p->in_gain + buffer[                       \
 200                        MOD(delay_pos + p->modulation_buffer[           \
 201                        modulation_pos],                                \
 202                        p->delay_buffer_length)] * p->decay;            \
 203                                                                        \
 204             modulation_pos = MOD(modulation_pos + 1,                   \
 205                              p->modulation_buffer_length);             \
 206             delay_pos = MOD(delay_pos + 1, p->delay_buffer_length);    \
 207             buffer[delay_pos] = v;                                     \
 208                                                                        \
 209             *d = v * p->out_gain;                                      \
 210         }                                                              \
 211     }                                                                  \
 212                                                                        \
 213     p->delay_pos      = delay_pos;                                     \
 214     p->modulation_pos = modulation_pos;                                \
 215 }
 216
 217 #define PHASER(name, type)                                              \
 218 static void phaser_## name (AudioPhaserContext *p,                      \
 219                             uint8_t * const *src, uint8_t **dst,        \
 220                             int nb_samples, int channels)               \
 221 {                                                                       \
 222     int i, c, delay_pos, modulation_pos;                                \
 223     type *s = (type *)src[0];                                           \
 224     type *d = (type *)dst[0];                                           \
 225     double *buffer = p->delay_buffer;                                   \
 226                                                                         \
 227     delay_pos      = p->delay_pos;                                      \
 228     modulation_pos = p->modulation_pos;                                 \
 229                                                                         \
 230     for (i = 0; i < nb_samples; i++) {                                  \
 231         int pos = MOD(delay_pos + p->modulation_buffer[modulation_pos], \
 232                    p->delay_buffer_length) * channels;                  \
 233         int npos;                                                       \
 234                                                                         \
 235         delay_pos = MOD(delay_pos + 1, p->delay_buffer_length);         \
 236         npos = delay_pos * channels;                                    \
 237         for (c = 0; c < channels; c++, s++, d++) {                      \
 238             double v = *s * p->in_gain + buffer[pos + c] * p->decay;    \
 239                                                                         \
 240             buffer[npos + c] = v;                                       \
 241                                                                         \
 242             *d = v * p->out_gain;                                       \
 243         }                                                               \
 244                                                                         \
 245         modulation_pos = MOD(modulation_pos + 1,                        \
 246                          p->modulation_buffer_length);                  \
 247     }                                                                   \
 248                                                                         \
 249     p->delay_pos      = delay_pos;                                      \
 250     p->modulation_pos = modulation_pos;                                 \
 251 }
 252
 253 PHASER_PLANAR(dbl, double)
 254 PHASER_PLANAR(flt, float)
 255 PHASER_PLANAR(s16, int16_t)
 256 PHASER_PLANAR(s32, int32_t)
 257
 258 PHASER(dbl, double)
 259 PHASER(flt, float)
 260 PHASER(s16, int16_t)
 261 PHASER(s32, int32_t)
 262
 263 static int config_output(AVFilterLink *outlink)
 264 {
 265     AudioPhaserContext *p = outlink->src->priv;
 266     AVFilterLink *inlink = outlink->src->inputs[0];
 267
 268     p->delay_buffer_length = p->delay * 0.001 * inlink->sample_rate + 0.5;
 269     p->delay_buffer = av_calloc(p->delay_buffer_length, sizeof(*p->delay_buffer) * inlink->channels);
 270     p->modulation_buffer_length = inlink->sample_rate / p->speed + 0.5;
 271     p->modulation_buffer = av_malloc(p->modulation_buffer_length * sizeof(*p->modulation_buffer));
 272
 273     if (!p->modulation_buffer || !p->delay_buffer)
 274         return AVERROR(ENOMEM);
 275
 276     generate_wave_table(p->type, AV_SAMPLE_FMT_S32,
 277                         p->modulation_buffer, p->modulation_buffer_length,
 278                         1., p->delay_buffer_length, M_PI / 2.0);
 279
 280     p->delay_pos = p->modulation_pos = 0;
 281
 282     switch (inlink->format) {
 283     case AV_SAMPLE_FMT_DBL:  p->phaser = phaser_dbl;  break;
 284     case AV_SAMPLE_FMT_DBLP: p->phaser = phaser_dblp; break;
 285     case AV_SAMPLE_FMT_FLT:  p->phaser = phaser_flt;  break;
 286     case AV_SAMPLE_FMT_FLTP: p->phaser = phaser_fltp; break;
 287     case AV_SAMPLE_FMT_S16:  p->phaser = phaser_s16;  break;
 288     case AV_SAMPLE_FMT_S16P: p->phaser = phaser_s16p; break;
 289     case AV_SAMPLE_FMT_S32:  p->phaser = phaser_s32;  break;
 290     case AV_SAMPLE_FMT_S32P: p->phaser = phaser_s32p; break;
 291     default: av_assert0(0);
 292     }
 293
 294     return 0;
 295 }
 296
 297 static int filter_frame(AVFilterLink *inlink, AVFrame *inbuf)
 298 {
 299     AudioPhaserContext *p = inlink->dst->priv;
 300     AVFilterLink *outlink = inlink->dst->outputs[0];
 301     AVFrame *outbuf;
 302
 303     if (av_frame_is_writable(inbuf)) {
 304         outbuf = inbuf;
 305     } else {
 306         outbuf = ff_get_audio_buffer(inlink, inbuf->nb_samples);
 307         if (!outbuf)
 308             return AVERROR(ENOMEM);
 309         av_frame_copy_props(outbuf, inbuf);
 310     }
 311
 312     p->phaser(p, inbuf->extended_data, outbuf->extended_data,
 313               outbuf->nb_samples, av_frame_get_channels(outbuf));
 314
 315     if (inbuf != outbuf)
 316         av_frame_free(&inbuf);
 317
 318     return ff_filter_frame(outlink, outbuf);
 319 }
 320
 321 static av_cold void uninit(AVFilterContext *ctx)
 322 {
 323     AudioPhaserContext *p = ctx->priv;
 324
 325     av_freep(&p->delay_buffer);
 326     av_freep(&p->modulation_buffer);
 327 }
 328
 329 static const AVFilterPad aphaser_inputs[] = {
 330     {
 331         .name         = "default",
 332         .type         = AVMEDIA_TYPE_AUDIO,
 333         .filter_frame = filter_frame,
 334     },
 335     { NULL }
 336 };
 337
 338 static const AVFilterPad aphaser_outputs[] = {
 339     {
 340         .name         = "default",
 341         .type         = AVMEDIA_TYPE_AUDIO,
 342         .config_props = config_output,
 343     },
 344     { NULL }
 345 };
 346
 347 AVFilter avfilter_af_aphaser = {
 348     .name          = "aphaser",
 349     .description   = NULL_IF_CONFIG_SMALL("Add a phasing effect to the audio."),
 350     .query_formats = query_formats,
 351     .priv_size     = sizeof(AudioPhaserContext),
 352     .init          = init,
 353     .uninit        = uninit,
 354     .inputs        = aphaser_inputs,
 355     .outputs       = aphaser_outputs,
 356     .priv_class    = &aphaser_class,
 357 };