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 #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     int type;
  41
  42     int delay_buffer_length;
  43     double *delay_buffer;
  44
  45     int modulation_buffer_length;
  46     int32_t *modulation_buffer;
  47
  48     int delay_pos, modulation_pos;
  49
  50     void (*phaser)(struct AudioPhaserContext *p,
  51                    uint8_t * const *src, uint8_t **dst,
  52                    int nb_samples, int channels);
  53 } AudioPhaserContext;
  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
  86 static int query_formats(AVFilterContext *ctx)
  87 {
  88     AVFilterFormats *formats;
  89     AVFilterChannelLayouts *layouts;
  90     static const enum AVSampleFormat sample_fmts[] = {
  91         AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBLP,
  92         AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP,
  93         AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32P,
  94         AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16P,
  95         AV_SAMPLE_FMT_NONE
  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)
 193 PHASER_PLANAR(s32, int32_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;
 208     p->modulation_buffer = av_malloc_array(p->modulation_buffer_length, sizeof(*p->modulation_buffer));
 209
 210     if (!p->modulation_buffer || !p->delay_buffer)
 211         return AVERROR(ENOMEM);
 212
 213     ff_generate_wave_table(p->type, AV_SAMPLE_FMT_S32,
 214                            p->modulation_buffer, p->modulation_buffer_length,
 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);
 263     av_freep(&p->modulation_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
 284 AVFilter ff_af_aphaser = {
 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 };