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 *s,
  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 *s = ctx->priv;
  77
  78     if (s->in_gain > (1 - s->decay * s->decay))
  79         av_log(ctx, AV_LOG_WARNING, "in_gain may cause clipping\n");
  80     if (s->in_gain / (1 - s->decay) > 1 / s->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     int ret;
  98
  99     layouts = ff_all_channel_counts();
 100     if (!layouts)
 101         return AVERROR(ENOMEM);
 102     ret = ff_set_common_channel_layouts(ctx, layouts);
 103     if (ret < 0)
 104         return ret;
 105
 106     formats = ff_make_format_list(sample_fmts);
 107     if (!formats)
 108         return AVERROR(ENOMEM);
 109     ret = ff_set_common_formats(ctx, formats);
 110     if (ret < 0)
 111         return ret;
 112
 113     formats = ff_all_samplerates();
 114     if (!formats)
 115         return AVERROR(ENOMEM);
 116     return ff_set_common_samplerates(ctx, formats);
 117 }
 118
 119 #define MOD(a, b) (((a) >= (b)) ? (a) - (b) : (a))
 120
 121 #define PHASER_PLANAR(name, type)                                      \
 122 static void phaser_## name ##p(AudioPhaserContext *s,                  \
 123                                uint8_t * const *ssrc, uint8_t **ddst,  \
 124                                int nb_samples, int channels)           \
 125 {                                                                      \
 126     int i, c, delay_pos, modulation_pos;                               \
 127                                                                        \
 128     av_assert0(channels > 0);                                          \
 129     for (c = 0; c < channels; c++) {                                   \
 130         type *src = (type *)ssrc[c];                                   \
 131         type *dst = (type *)ddst[c];                                   \
 132         double *buffer = s->delay_buffer +                             \
 133                          c * s->delay_buffer_length;                   \
 134                                                                        \
 135         delay_pos      = s->delay_pos;                                 \
 136         modulation_pos = s->modulation_pos;                            \
 137                                                                        \
 138         for (i = 0; i < nb_samples; i++, src++, dst++) {               \
 139             double v = *src * s->in_gain + buffer[                     \
 140                        MOD(delay_pos + s->modulation_buffer[           \
 141                        modulation_pos],                                \
 142                        s->delay_buffer_length)] * s->decay;            \
 143                                                                        \
 144             modulation_pos = MOD(modulation_pos + 1,                   \
 145                              s->modulation_buffer_length);             \
 146             delay_pos = MOD(delay_pos + 1, s->delay_buffer_length);    \
 147             buffer[delay_pos] = v;                                     \
 148                                                                        \
 149             *dst = v * s->out_gain;                                    \
 150         }                                                              \
 151     }                                                                  \
 152                                                                        \
 153     s->delay_pos      = delay_pos;                                     \
 154     s->modulation_pos = modulation_pos;                                \
 155 }
 156
 157 #define PHASER(name, type)                                              \
 158 static void phaser_## name (AudioPhaserContext *s,                      \
 159                             uint8_t * const *ssrc, uint8_t **ddst,      \
 160                             int nb_samples, int channels)               \
 161 {                                                                       \
 162     int i, c, delay_pos, modulation_pos;                                \
 163     type *src = (type *)ssrc[0];                                        \
 164     type *dst = (type *)ddst[0];                                        \
 165     double *buffer = s->delay_buffer;                                   \
 166                                                                         \
 167     delay_pos      = s->delay_pos;                                      \
 168     modulation_pos = s->modulation_pos;                                 \
 169                                                                         \
 170     for (i = 0; i < nb_samples; i++) {                                  \
 171         int pos = MOD(delay_pos + s->modulation_buffer[modulation_pos], \
 172                       s->delay_buffer_length) * channels;               \
 173         int npos;                                                       \
 174                                                                         \
 175         delay_pos = MOD(delay_pos + 1, s->delay_buffer_length);         \
 176         npos = delay_pos * channels;                                    \
 177         for (c = 0; c < channels; c++, src++, dst++) {                  \
 178             double v = *src * s->in_gain + buffer[pos + c] * s->decay;  \
 179                                                                         \
 180             buffer[npos + c] = v;                                       \
 181                                                                         \
 182             *dst = v * s->out_gain;                                     \
 183         }                                                               \
 184                                                                         \
 185         modulation_pos = MOD(modulation_pos + 1,                        \
 186                          s->modulation_buffer_length);                  \
 187     }                                                                   \
 188                                                                         \
 189     s->delay_pos      = delay_pos;                                      \
 190     s->modulation_pos = modulation_pos;                                 \
 191 }
 192
 193 PHASER_PLANAR(dbl, double)
 194 PHASER_PLANAR(flt, float)
 195 PHASER_PLANAR(s16, int16_t)
 196 PHASER_PLANAR(s32, int32_t)
 197
 198 PHASER(dbl, double)
 199 PHASER(flt, float)
 200 PHASER(s16, int16_t)
 201 PHASER(s32, int32_t)
 202
 203 static int config_output(AVFilterLink *outlink)
 204 {
 205     AudioPhaserContext *s = outlink->src->priv;
 206     AVFilterLink *inlink = outlink->src->inputs[0];
 207
 208     s->delay_buffer_length = s->delay * 0.001 * inlink->sample_rate + 0.5;
 209     if (s->delay_buffer_length <= 0) {
 210         av_log(outlink->src, AV_LOG_ERROR, "delay is too small\n");
 211         return AVERROR(EINVAL);
 212     }
 213     s->delay_buffer = av_calloc(s->delay_buffer_length, sizeof(*s->delay_buffer) * inlink->channels);
 214     s->modulation_buffer_length = inlink->sample_rate / s->speed + 0.5;
 215     s->modulation_buffer = av_malloc_array(s->modulation_buffer_length, sizeof(*s->modulation_buffer));
 216
 217     if (!s->modulation_buffer || !s->delay_buffer)
 218         return AVERROR(ENOMEM);
 219
 220     ff_generate_wave_table(s->type, AV_SAMPLE_FMT_S32,
 221                            s->modulation_buffer, s->modulation_buffer_length,
 222                            1., s->delay_buffer_length, M_PI / 2.0);
 223
 224     s->delay_pos = s->modulation_pos = 0;
 225
 226     switch (inlink->format) {
 227     case AV_SAMPLE_FMT_DBL:  s->phaser = phaser_dbl;  break;
 228     case AV_SAMPLE_FMT_DBLP: s->phaser = phaser_dblp; break;
 229     case AV_SAMPLE_FMT_FLT:  s->phaser = phaser_flt;  break;
 230     case AV_SAMPLE_FMT_FLTP: s->phaser = phaser_fltp; break;
 231     case AV_SAMPLE_FMT_S16:  s->phaser = phaser_s16;  break;
 232     case AV_SAMPLE_FMT_S16P: s->phaser = phaser_s16p; break;
 233     case AV_SAMPLE_FMT_S32:  s->phaser = phaser_s32;  break;
 234     case AV_SAMPLE_FMT_S32P: s->phaser = phaser_s32p; break;
 235     default: av_assert0(0);
 236     }
 237
 238     return 0;
 239 }
 240
 241 static int filter_frame(AVFilterLink *inlink, AVFrame *inbuf)
 242 {
 243     AudioPhaserContext *s = inlink->dst->priv;
 244     AVFilterLink *outlink = inlink->dst->outputs[0];
 245     AVFrame *outbuf;
 246
 247     if (av_frame_is_writable(inbuf)) {
 248         outbuf = inbuf;
 249     } else {
 250         outbuf = ff_get_audio_buffer(outlink, inbuf->nb_samples);
 251         if (!outbuf) {
 252             av_frame_free(&inbuf);
 253             return AVERROR(ENOMEM);
 254         }
 255         av_frame_copy_props(outbuf, inbuf);
 256     }
 257
 258     s->phaser(s, inbuf->extended_data, outbuf->extended_data,
 259               outbuf->nb_samples, outbuf->channels);
 260
 261     if (inbuf != outbuf)
 262         av_frame_free(&inbuf);
 263
 264     return ff_filter_frame(outlink, outbuf);
 265 }
 266
 267 static av_cold void uninit(AVFilterContext *ctx)
 268 {
 269     AudioPhaserContext *s = ctx->priv;
 270
 271     av_freep(&s->delay_buffer);
 272     av_freep(&s->modulation_buffer);
 273 }
 274
 275 static const AVFilterPad aphaser_inputs[] = {
 276     {
 277         .name         = "default",
 278         .type         = AVMEDIA_TYPE_AUDIO,
 279         .filter_frame = filter_frame,
 280     },
 281     { NULL }
 282 };
 283
 284 static const AVFilterPad aphaser_outputs[] = {
 285     {
 286         .name         = "default",
 287         .type         = AVMEDIA_TYPE_AUDIO,
 288         .config_props = config_output,
 289     },
 290     { NULL }
 291 };
 292
 293 const AVFilter ff_af_aphaser = {
 294     .name          = "aphaser",
 295     .description   = NULL_IF_CONFIG_SMALL("Add a phasing effect to the audio."),
 296     .query_formats = query_formats,
 297     .priv_size     = sizeof(AudioPhaserContext),
 298     .init          = init,
 299     .uninit        = uninit,
 300     .inputs        = aphaser_inputs,
 301     .outputs       = aphaser_outputs,
 302     .priv_class    = &aphaser_class,
 303 };